Update for 13.13.0-rc2

This commit is contained in:
Kevin Harwell
2016-11-22 13:02:41 -05:00
parent e9268b3b58
commit 89050e710a
6 changed files with 135 additions and 1064 deletions

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13.13.0 13.13.0-rc2

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2016-11-22 17:20 +0000 Asterisk Development Team <asteriskteam@digium.com> 2016-11-22 18:02 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.13.0 Released. * asterisk 13.13.0-rc2 Released.
2016-11-21 09:40 +0000 [e246b36a3c] gtjoseph <gjoseph@digium.com>
* build: Backport addition of librt check to configure.ac
A while back, a master-only change was made to check for librt which
should probably have been cherry-picked to 13 at that time. Sometime
between then and now, part of that change did make it into 13 but it
was incomplete and non-functional. This patch backports the rest
of the librt check and allows the link of libasteriskpj to use the
results.
Change-Id: I1424008fd8c90f389dda53162ec4a340b253a3c1
2016-11-22 11:20 +0000 [855f05e525] Kevin Harwell <kharwell@digium.com>
* Update for 13.13.0
2016-11-18 18:59 +0000 Asterisk Development Team <asteriskteam@digium.com> 2016-11-18 18:59 +0000 Asterisk Development Team <asteriskteam@digium.com>

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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-13.13.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.13.0-rc2</h3><h3 align="center">Date: 2016-11-22</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.13.0-rc1.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">1 gtjoseph <gjoseph@digium.com><br/>1 Kevin Harwell <kharwell@digium.com><br/></td><td width="33%"><td width="33%"></tr>
</table><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e246b36a3c1f178e57e27c6a5fc40eba5d364c9b">e246b36a3c</a></td><td>gtjoseph</td><td>build: Backport addition of librt check to configure.ac</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=855f05e525e81012638aed3366559de3c1c9eb86">855f05e525</a></td><td>Kevin Harwell</td><td>Update for 13.13.0</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-13.13.0-rc1-summary.html | 308 --------------
asterisk-13.13.0-rc1-summary.txt | 764 -------------------------------------
b/.version | 2
b/ChangeLog | 4
b/asterisk-13.13.0-summary.html | 301 ++++++++++++++
b/asterisk-13.13.0-summary.txt | 760 ++++++++++++++++++++++++++++++++++++
b/configure | 194 ++++++++-
b/configure.ac | 4
b/include/asterisk/autoconfig.h.in | 3
b/main/Makefile | 1
10 files changed, 1249 insertions(+), 1092 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.13.0-rc2
Date: 2016-11-22
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Other Changes
4. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-13.13.0-rc1.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were affected by commits that went into
this release.
Coders Testers Reporters
1 gtjoseph
1 Kevin Harwell
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+---------------+-------------------------------------------|
| e246b36a3c | gtjoseph | build: Backport addition of librt check |
| | | to configure.ac |
|------------+---------------+-------------------------------------------|
| 855f05e525 | Kevin Harwell | Update for 13.13.0 |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
asterisk-13.13.0-rc1-summary.html | 308 --------------
asterisk-13.13.0-rc1-summary.txt | 764 -------------------------------------
b/.version | 2
b/ChangeLog | 4
b/asterisk-13.13.0-summary.html | 301 ++++++++++++++
b/asterisk-13.13.0-summary.txt | 760 ++++++++++++++++++++++++++++++++++++
b/configure | 194 ++++++++-
b/configure.ac | 4
b/include/asterisk/autoconfig.h.in | 3
b/main/Makefile | 1
10 files changed, 1249 insertions(+), 1092 deletions(-)

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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-13.13.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.13.0</h3><h3 align="center">Date: 2016-11-22</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#open_issues">Open Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.12.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">20 gtjoseph <gjoseph@digium.com><br/>11 Joshua Colp <jcolp@digium.com><br/>10 Matt Jordan <mjordan@digium.com><br/>10 Mark Michelson <mmichelson@digium.com><br/>7 Richard Mudgett <rmudgett@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>3 Alexander Traud <pabstraud@compuserve.com><br/>3 Corey Farrell <git@cfware.com><br/>3 Alexander Anikin <may213@yandex.ru><br/>2 Sebastian Gutierrez <sgutierrez@integraccs.com><br/>1 Michael Walton <mike@farsouthnet.com><br/>1 Etienne Lessard <elessard@proformatique.com><br/>1 Leandro Dardini <ldardini@gmail.com><br/>1 snuffy <snuffy22@gmail.com><br/>1 Pascal Cadotte Michaud <pcadotte@proformatique.com><br/>1 Maciej Szmigiero <mail@maciej.szmigiero.name><br/>1 Michael Kuron <m.kuron@gmx.de><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Grachev Sergey <grachev@mcn.ru><br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Igor Goncharovskiy <igor.goncharovsky@gmail.com><br/>1 Moises Silva <moises.silva@gmail.com><br/></td><td width="33%">1 Dmitry Melekhov<br/></td><td width="33%">7 Matt Jordan <mjordan@digium.com><br/>5 Alexander Traud <pabstraud@compuserve.com><br/>4 Morten Tryfoss <morten@tryfoss.no><br/>4 scgm11 <scgm11@gmail.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 George Joseph <gjoseph@digium.com><br/>3 Richard Mudgett <rmudgett@digium.com><br/>2 Gabriele Giacone <1o5g4r8o@gmail.com><br/>2 Andrew Nagy <andrew.nagy@the159.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Dmitry Melekhov<br/>1 Andreas Wetzel <mickey242@gmx.net><br/>1 Ian Gilmour<br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Bill Brigden <bill@brigden.me><br/>1 Andrew Nagy<br/>1 Sergey Grachev <FreeSS@live.ru><br/>1 snuffy <snuffy22@gmail.com><br/>1 Daniele Pallastrelli <daniele.pallastrelli@sadel.it><br/>1 Kevin Harwell <kharwell@digium.com><br/>1 Kayode <kayode.olajide@gltd.net><br/>1 Michael Keuter <lists@mksolutions.info><br/>1 Dmitry Melekhov <dm@belkam.com><br/>1 Harley Peters <harley@thepetersclan.com><br/>1 Corey Farrell <git@cfware.com><br/>1 Leandro Dardini <ldardini@gmail.com><br/>1 Jonathan Harris <lardconcepts@gmail.com><br/>1 Badalian Vyacheslav <slavon.net@gmail.com><br/>1 Doug Lytle <support@drdos.info><br/>1 scgm11<br/>1 Richard Mudgett<br/>1 Maciej Szmigiero <mail@maciej.szmigiero.name><br/>1 Etienne Lessard <elessard@proformatique.com><br/>1 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>1 John Kiniston <johnkiniston@gmail.com><br/>1 Jason <asterisk@srpl.com><br/>1 Florian Loyau <florian.loyau@astrium-eu-projects.eu><br/>1 Ian Gilmour <ian.gilmour.x@gmail.com><br/>1 Michael Walton <mike@farsouthnet.com><br/>1 Morton Tryfoss<br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>New Feature</h3><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26595">ASTERISK-26595</a>: ARI: Add the ability to control the source of video in a multi-party mixing bridge<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d23b4af4779675589c8a3ce39c0f4b80d0432d5c">[d23b4af477]</a> Matt Jordan -- res/ari/resource_bridges: Add the ability to manipulate the video source</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26470">ASTERISK-26470</a>: ARI: Add an 'asterisk_id' field to outgoing events<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42cfdcd1b700af157746b897dc04362e853065c0">[42cfdcd1b7]</a> Matt Jordan -- res/ari: Add the Asterisk EID field to outgoing events</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24400">ASTERISK-24400</a>: ooh323 sends wrong hangup code<br/>Reported by: Dmitry Melekhov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9ac1f5de474feee8933530b6370acdf3a45cc3f">[a9ac1f5de4]</a> Alexander Anikin -- chan_ooh323: Fixes to work right with Cisco devices</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26549">ASTERISK-26549</a>: app_dial: When PickupChan() is used some channels may have incorrect device state<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d971647949a5a6dee5c80526f2baa90b02687ad5">[d971647949]</a> Joshua Colp -- app_dial: Fix incorrect device state when channel is picked up.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26462">ASTERISK-26462</a>: [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage<br/>Reported by: Leandro Dardini<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0306869399c519fd6f1b35332aa639a8db4b86fb">[0306869399]</a> Leandro Dardini -- app_queue: Added initialization for "context" parameter</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26503">ASTERISK-26503</a>: app_voicemail: Asterisk crashes when MailboxExists is used<br/>Reported by: Doug Lytle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14496ce1e5bbc9b2192ca8f882ecb99887c5d40a">[14496ce1e5]</a> Joshua Colp -- app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.</li>
</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26555">ASTERISK-26555</a>: Multi-party Video: Fix some post Asterisk-11 regressions<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7dc536b7ae11abf0414b367d1125610dd858188">[e7dc536b7a]</a> Matt Jordan -- main/bridge_channel: Fix channel reference leak on video source</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c824b955d4200c82805b3e07aa3af30c43fd09d">[7c824b955d]</a> Matt Jordan -- main/bridge: Add some verbose logging for video source changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd6af2dee8016ac64b3c4943d8ded04767a0a752">[fd6af2dee8]</a> Matt Jordan -- bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26412">ASTERISK-26412</a>: build: Prepare for gcc 6.2<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd4d7d8ad05bbe8d86b89429dfa55fbc3a0c108d">[bd4d7d8ad0]</a> Kevin Harwell -- stasis_recording/stored: remove calls to deprecated readdir_r function.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26516">ASTERISK-26516</a>: pjsip: Memory corruption with possible memory leak.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e632222bc41d860af7de2463c35de60387a2f295">[e632222bc4]</a> Richard Mudgett -- res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afecb2cfc084db522cf570aa2210056b1c396196">[afecb2cfc0]</a> Richard Mudgett -- bundled pjproject: Fix DNS write to freed memory.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26444">ASTERISK-26444</a>: 'features show' command in CLI does not return prompt.<br/>Reported by: John Kiniston<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2036c827cb22e2fbf509d4318b6f177d516c033">[c2036c827c]</a> snuffy -- Fix issue with CLI not returning to prompt after running "features show"</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26482">ASTERISK-26482</a>: [patch] chan_pjsip: segfault on already disconnected session<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d462b9eaf4f34a518ec491f1f97c21adc30a87c">[6d462b9eaf]</a> Alexei Gradinari -- chan_pjsip: segfault on already disconnected session</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26523">ASTERISK-26523</a>: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression<br/>Reported by: Michael Keuter<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb30963d222cb1e12af9bbf8dfed58001c9fcaf4">[cb30963d22]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26476">ASTERISK-26476</a>: chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"<br/>Reported by: Sergey Grachev<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3f10b7b94c08e9024057d0237a83da1bda6a946">[b3f10b7b94]</a> Grachev Sergey -- chan_sip: Incorrect display option Outbound reg. retry 403</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26457">ASTERISK-26457</a>: [patch] force_rport,auto_comedia: No NAT detection triggered.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a859bcb49cc2c60ff2979853aae8c54269287598">[a859bcb49c]</a> Alexander Traud -- chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia.</li>
</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26565">ASTERISK-26565</a>: chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set<br/>Reported by: Jason<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3faca1d4ffb3e727b792958f3b124f2258643259">[3faca1d4ff]</a> Igor Goncharovskiy -- Fix closing rtp ports after call finished in chan_unistim.</li>
</ul><br><h4>Category: Codecs/codec_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26520">ASTERISK-26520</a>: codec_opus: Generated fmtp line has no content<br/>Reported by: scgm11<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c031b67d3c7da83bab914bcdaad45d2a0bc9ff8">[2c031b67d3]</a> Mark Michelson -- res_format_attr_opus: Fix fmtp generation.</li>
</ul><br><h4>Category: Core/AstMM</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26526">ASTERISK-26526</a>: [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=30b1bc77d2cfea0a593ecead8e392e468b40430c">[30b1bc77d2]</a> Corey Farrell -- vector: Prevent NULL argument to memcpy.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26524">ASTERISK-26524</a>: astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b96f18560b529b614d0773a060bc03ef73498c61">[b96f18560b]</a> Corey Farrell -- astobj2: Declare private variable data_size for AO2_DEBUG only.</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26555">ASTERISK-26555</a>: Multi-party Video: Fix some post Asterisk-11 regressions<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7dc536b7ae11abf0414b367d1125610dd858188">[e7dc536b7a]</a> Matt Jordan -- main/bridge_channel: Fix channel reference leak on video source</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c824b955d4200c82805b3e07aa3af30c43fd09d">[7c824b955d]</a> Matt Jordan -- main/bridge: Add some verbose logging for video source changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd6af2dee8016ac64b3c4943d8ded04767a0a752">[fd6af2dee8]</a> Matt Jordan -- bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26608">ASTERISK-26608</a>: Compile and link failures on OpenBSD<br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b213045fe4a32d4b41ca9a29af383b17a885ca35">[b213045fe4]</a> gtjoseph -- build: Various OpenBSD issues</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26592">ASTERISK-26592</a>: Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e0c22404316ecdf8e1510553474274eddf55e20">[5e0c224043]</a> gtjoseph -- cli: Fix ast_el_read_char to work with libedit >= 3.1</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22480">ASTERISK-22480</a>: Embedded pjproject: build.mak contains hardcoded full path to version.mak<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61a5c3460ec23a623ac62633d055b34d4dded682">[61a5c3460e]</a> gtjoseph -- pjproject_bundled: Remove usage of tar's --strip-components option</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26356">ASTERISK-26356</a>: menuselect: invalid test for GTK2<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6f5880913f80372dd042ae4c9e874d9eab840f55">[6f5880913f]</a> Tzafrir Cohen -- menuselect: invalid test for GTK2</li>
</ul><br><h4>Category: Core/CodecInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26605">ASTERISK-26605</a>: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed0f1afc8cea8918171a1aabc9b6885bba41e3c4">[ed0f1afc8c]</a> Richard Mudgett -- codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26605">ASTERISK-26605</a>: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed0f1afc8cea8918171a1aabc9b6885bba41e3c4">[ed0f1afc8c]</a> Richard Mudgett -- codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26509">ASTERISK-26509</a>: A few non-critical deprecation warnings when building on Ubuntu 16.10<br/>Reported by: Jonathan Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd4d7d8ad05bbe8d86b89429dfa55fbc3a0c108d">[bd4d7d8ad0]</a> Kevin Harwell -- stasis_recording/stored: remove calls to deprecated readdir_r function.</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26537">ASTERISK-26537</a>: AMI: NewConnectedLine event is not documented<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42bd70b29f5673ffead10c70cc4096c1410f4144">[42bd70b29f]</a> Etienne Lessard -- manager: Add documentation for NewConnectedLine event.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26311">ASTERISK-26311</a>: [patch] rtp_engine: Allow more than 32 dynamic payload types.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0cf1778eed4754570a36938e1f5d212951320a71">[0cf1778eed]</a> Alexander Traud -- rtp_engine: Allow more than 32 dynamic payload types.</li>
</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26468">ASTERISK-26468</a>: ari: Bridge events stop working after this sequence of ARI calls<br/>Reported by: Daniele Pallastrelli<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a1f9c5dab3fbb2159575d74721aa0f4ddc3d078">[3a1f9c5dab]</a> Joshua Colp -- res_stasis: Don't unsubscribe from a NULL bridge.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26514">ASTERISK-26514</a>: Super Awesome Company: Don't specify transport in pjsip.conf<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87903a684888f88a46746801262e7f14a2249d01">[87903a6848]</a> Rusty Newton -- SAC documentation: don't specify transports for endpoints and registrations</li>
</ul><br><h4>Category: Features</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26444">ASTERISK-26444</a>: 'features show' command in CLI does not return prompt.<br/>Reported by: John Kiniston<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2036c827cb22e2fbf509d4318b6f177d516c033">[c2036c827c]</a> snuffy -- Fix issue with CLI not returning to prompt after running "features show"</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26575">ASTERISK-26575</a>: testsuite: Need to check PJSIP functionality when res_srtp is not loaded.<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b70eb07c53d041e868ced079759471220f78bf50">[b70eb07c53]</a> Joshua Colp -- res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25070">ASTERISK-25070</a>: Fix FTBFS on Hurd<br/>Reported by: Gabriele Giacone<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94c9496ed5a73a83ac26c58f8ae049acccc0fd51">[94c9496ed5]</a> Tzafrir Cohen -- netsock.c: fix includes for HURD</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1c9487375f4dc80c5425a9ccfd407f3fa849ac3">[c1c9487375]</a> Tzafrir Cohen -- define PATH_MAX for HURD</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26387">ASTERISK-26387</a>: Asterisk segfaults shortly after starting even with no active calls. <br/>Reported by: Harley Peters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d7f457e4c15235e33add6eb66158df0fbd9bf0b5">[d7f457e4c1]</a> Richard Mudgett -- bundled pjproject: Crashes while resolving DNS names.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26513">ASTERISK-26513</a>: tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f373de302032c13487cfcaa616fc070f10d68b57">[f373de3020]</a> Corey Farrell -- Fix shutdown crash caused by modules being left open.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26421">ASTERISK-26421</a>: Segmentation Fault with ARI originate into mixing bridge with 43 clients<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eff97808fb95e4f9de13c90990f8ef5435352f31">[eff97808fb]</a> Mark Michelson -- ARI: Detect duplicate channel IDs</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=012fda29d23bac1d2b06e9a8933980047da30246">[012fda29d2]</a> Mark Michelson -- CDR: Alter destruction pattern for CDR chains.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26480">ASTERISK-26480</a>: [patch] CLI: core set debug: Auto-completes File not Module<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=662b560c3531d8b70bdf8b91e68ae965926663cd">[662b560c35]</a> Alexander Traud -- cli: Auto-complete File not Module for core set debug.</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26343">ASTERISK-26343</a>: ASTERISK-25951 causes issues for callerid manipulation through agi<br/>Reported by: Morten Tryfoss<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=404596b79017611c47cfd2e260207ace1dbb9208">[404596b790]</a> gtjoseph -- channel: Fix issues in hangup scenarios caused by frame deferral</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2e3a3545754749de21873bfdc6d1a40ec7d8893f">[2e3a354575]</a> Mark Michelson -- autoservice: Use frame deferral API</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c10091f3d1430c6fc04015226f8c3e3aa9d8282">[5c10091f3d]</a> Mark Michelson -- AGI: Only defer frames when in an interception routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9231a56cf3d6f5eca1bf2d37d827453400690773">[9231a56cf3]</a> Mark Michelson -- Add API for channel frame deferral.</li>
</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26468">ASTERISK-26468</a>: ari: Bridge events stop working after this sequence of ARI calls<br/>Reported by: Daniele Pallastrelli<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a1f9c5dab3fbb2159575d74721aa0f4ddc3d078">[3a1f9c5dab]</a> Joshua Colp -- res_stasis: Don't unsubscribe from a NULL bridge.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26516">ASTERISK-26516</a>: pjsip: Memory corruption with possible memory leak.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e632222bc41d860af7de2463c35de60387a2f295">[e632222bc4]</a> Richard Mudgett -- res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afecb2cfc084db522cf570aa2210056b1c396196">[afecb2cfc0]</a> Richard Mudgett -- bundled pjproject: Fix DNS write to freed memory.</li>
</ul><br><h4>Category: Resources/res_pjsip_caller_id</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26307">ASTERISK-26307</a>: res_pjsip_caller_id: Crash on outgoing change<br/>Reported by: Bill Brigden<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=675c71ae8ca9fc086a5da21a5abcb7ef72506343">[675c71ae8c]</a> Joshua Colp -- res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26541">ASTERISK-26541</a>: res_pjsip_sdp_rtp: Restrict number of formats to maximum<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f188bb7a84b5cb065f35d6068b0c800a695a940">[5f188bb7a8]</a> Joshua Colp -- res_pjsip_sdp_rtp: Limit number of formats to defined maximum.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26423">ASTERISK-26423</a>: res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness<br/>Reported by: Andreas Wetzel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0bc17edfff27bb9dbbe931814fb5653005f3219">[e0bc17edff]</a> Joshua Colp -- pjsip: Fix a few media bugs with reinvites and asymmetric payloads.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26309">ASTERISK-26309</a>: [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f534f67f520397079f1af54358aa4404403025eb">[f534f67f52]</a> Joshua Colp -- res_pjsip_sdp_rtp: Fix address family of explicit media_address.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb982480d815fb0d5059fbfa86682cd30846556c">[bb982480d8]</a> Joshua Colp -- pjsip: Support dual stack automatically.</li>
</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26510">ASTERISK-26510</a>: pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61a5c3460ec23a623ac62633d055b34d4dded682">[61a5c3460e]</a> gtjoseph -- pjproject_bundled: Remove usage of tar's --strip-components option</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26344">ASTERISK-26344</a>: Asterisk 13.11.0 + PJSIP crash<br/>Reported by: Ian Gilmour<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d7f457e4c15235e33add6eb66158df0fbd9bf0b5">[d7f457e4c1]</a> Richard Mudgett -- bundled pjproject: Crashes while resolving DNS names.</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26558">ASTERISK-26558</a>: app_queue: add variable to know if the call is not answered after a queue<br/>Reported by: scgm11<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71dc33356504e32b6ed4bbf6faaacb51a3602d10">[71dc333565]</a> Joshua Colp -- app_queue: Add mention of 'ABANDON' variable to CHANGES.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fd5031c1cac30f7dacea803a8d9e07302edcbf7">[7fd5031c1c]</a> Sebastian Gutierrez -- app_queue: new variable set when abandoned</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26176">ASTERISK-26176</a>: chan_sip: Add AccountCode to AMI PeerEntry<br/>Reported by: scgm11<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=714412f6c43f2b6b8f8af5a7d0d2bb337a24701c">[714412f6c4]</a> Sebastian Gutierrez -- chan_sip: add missing account code</li>
</ul><br><h4>Category: Codecs/codec_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26538">ASTERISK-26538</a>: codec_opus: Add sample to configs/samples/codecs.conf.sample<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50fa868ab81992dfdfc0f50834fa06cb370290c3">[50fa868ab8]</a> Kevin Harwell -- codecs.conf.sample: Add sample and option descriptions for codec_opus</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25063">ASTERISK-25063</a>: [patch]add X.509 subject alternative name support to Asterisk TLS support<br/>Reported by: Maciej Szmigiero<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b96e8cc3d3db3d30921905203520b4e08b527b8">[7b96e8cc3d]</a> Maciej Szmigiero -- Add X.509 subject alternative name support to TLS certificate</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26488">ASTERISK-26488</a>: ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29692d4aa42bc1ce9b737dd0fa4911a2bf5c3ce3">[29692d4aa4]</a> Matt Jordan -- res/stasis: Add CLI commands for displaying/debugging ARI apps</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26418">ASTERISK-26418</a>: res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c62b60e56e5c06ae7d33d81d03268d0d2f6aa28">[3c62b60e56]</a> Michael Walton -- res_rtp_asterisk: Add ice_blacklist option</li>
</ul><br><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Channels/chan_multicast_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26439">ASTERISK-26439</a>: chan_rtp: Crash when originating<br/>Reported by: Kayode<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=644fad74770594c10b1819900d6195855068020a">[644fad7477]</a> Moises Silva -- chan_rtp: Set a sane default rtp engine for unicast.</li>
</ul><br><h4>Category: Core/Jitterbuffer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25270">ASTERISK-25270</a>: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind<br/>Reported by: Florian Loyau<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb30963d222cb1e12af9bbf8dfed58001c9fcaf4">[cb30963d22]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25270">ASTERISK-25270</a>: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind<br/>Reported by: Florian Loyau<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb30963d222cb1e12af9bbf8dfed58001c9fcaf4">[cb30963d22]</a> Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"</li>
</ul><br><h4>Category: Resources/res_rtp_multicast</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26439">ASTERISK-26439</a>: chan_rtp: Crash when originating<br/>Reported by: Kayode<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=644fad74770594c10b1819900d6195855068020a">[644fad7477]</a> Moises Silva -- chan_rtp: Set a sane default rtp engine for unicast.</li>
</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=751d43e8e4173386be5455311561dfa819a642d3">751d43e8e4</a></td><td>Joshua Colp</td><td>Update for 13.13.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb624b10ae55450ead02f67dbcfb918f3eb2e372">cb624b10ae</a></td><td>Mark Michelson</td><td>Bump ARI version to 1.10.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bde3d022a3663e69e1da003f0ed1665a6eabd3fd">bde3d022a3</a></td><td>Mark Michelson</td><td>manager: update minor version</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c92dcc76da6eb57a9c9b9701c6577b0f8dde578f">c92dcc76da</a></td><td>gtjoseph</td><td>file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0790aa528a902b49de9aff8232ff4c7a97ea6fda">0790aa528a</a></td><td>Matt Jordan</td><td>pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=993a6f96c7f6e07849af4a923ad3a0497ca854bd">993a6f96c7</a></td><td>Matt Jordan</td><td>apps/app_echo: Only relay a single video source change frame</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=404a62eeeecec486065e45aaaf5a0d58b0adec0a">404a62eeee</a></td><td>gtjoseph</td><td>Revert "Revert "channel: Use frame deferral API for safe sleep.""</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09d8febc91bc24a9c66c65616cf9a2502a501dcb">09d8febc91</a></td><td>gtjoseph</td><td>Revert "Revert "autoservice: Use frame deferral API""</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ffad2b44dfca5b3ced0265a085ddc67dfb0f86a6">ffad2b44df</a></td><td>gtjoseph</td><td>Revert "Revert "AGI: Only defer frames when in an interception routine.""</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fefb6187ffe6d92b863ca3c2ee20a4a51554498">2fefb6187f</a></td><td>gtjoseph</td><td>Revert "Revert "Add API for channel frame deferral.""</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=412d43fa213fb7677561e0b0f842b56550440d8c">412d43fa21</a></td><td>Richard Mudgett</td><td>res_pjsip.c: Rework endpt_send_request() req_wrapper code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2e7fc56d3cb55b0fe200c0cb8f183e94698cfabc">2e7fc56d3c</a></td><td>Richard Mudgett</td><td>res_pjsip: Fix tdata leaks in off nominal paths.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da68b185b3469361c33b47d93455624aac2e5e5d">da68b185b3</a></td><td>Richard Mudgett</td><td>res_pjsip_registrar_expire.c: Remove extra linefeed in debug message.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b5a7ced136b7178ae0b2ba39221eba1cd2e37c9">6b5a7ced13</a></td><td>gtjoseph</td><td>Revert "Add API for channel frame deferral."</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6be5d8de0da7e804544507f70382425af9a07b3f">6be5d8de0d</a></td><td>gtjoseph</td><td>Revert "AGI: Only defer frames when in an interception routine."</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1df434e2b4bd7cc34b9b4addf405a3caa7ac16b8">1df434e2b4</a></td><td>gtjoseph</td><td>Revert "autoservice: Use frame deferral API"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=58c88cfbaa80cb43419cde9186d643d1c5d24baf">58c88cfbaa</a></td><td>gtjoseph</td><td>Revert "channel: Use frame deferral API for safe sleep."</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a562fbe618e0e3b034e7006314b4ea1dfef6d732">a562fbe618</a></td><td>gtjoseph</td><td>build: Fix default values for some SANITIZER options</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e043d1a55cf356066b3b39ebac8b4bbb612ce807">e043d1a55c</a></td><td>Mark Michelson</td><td>res_pjsip_session: Do not call session supplements when it's too late.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44f7e252397fd87420b3374df26941d7436401b3">44f7e25239</a></td><td>Mark Michelson</td><td>channel: Use frame deferral API for safe sleep.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0ee249075a1433a9afc028d0de82dd9f7748025f">0ee249075a</a></td><td>Alexander Anikin</td><td>chan_ooh323: reset rrq count on gk registration</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59c23e1768a48825cd628b8c4b59f7eb46b8d364">59c23e1768</a></td><td>Michael Kuron</td><td>automon: restore mixing of the both channels after recording stops</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e79acaeb750ba286786ad122ddf20e9c179c19cf">e79acaeb75</a></td><td>Matt Jordan</td><td>res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a831969855c3117fa086a9ece9165e04cd1b2d4">7a83196985</a></td><td>Matt Jordan</td><td>res_stasis: Set a video source mode on Stasis created bridges</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eceab15f3339445c37b756c9dbe338f95850741f">eceab15f33</a></td><td>Alexander Anikin</td><td>chan_ooh323: Fix infinite loop on read second part of H.225 packet</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9992da4aaa4807bac8384ff63e3b3e576d5ad3a">a9992da4aa</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix issue with libasteriskpj needing libresample</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a36a7d0cf4e73bd9f934e063ee5e4dc057921744">a36a7d0cf4</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix compile of pjsua so it handles audio</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1c55dc9be90904363dbcf958a9fcc243c85629">6b1c55dc9b</a></td><td>gtjoseph</td><td>pjproject_bundled: Fix issue where "/version.mak" wasn't found</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a2092b72266b78ab2c10bfac03bf28f207becdc">3a2092b722</a></td><td>gtjoseph</td><td>test_astobj2_thrash: Fix multithreaded issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=640203802ee01354f5aacfa13cb035dfeb681b4c">640203802e</a></td><td>Pascal Cadotte Michaud</td><td>typo: s/paranthesis/parenthesis/ in a comment</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9b3557e0543233ce8dad27f3718a02025d1212b1">9b3557e054</a></td><td>gtjoseph</td><td>pjproject_bundled: Fixed various build issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74d9385273e914e3349cab63b7e478a0956bcf8a">74d9385273</a></td><td>gtjoseph</td><td>utils.c: Fix ast_set_default_eid for multiple platforms</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-13.12.0-summary.html | 543 ----
asterisk-13.12.0-summary.txt | 1275 ---------
b/.version | 2
b/CHANGES | 95
b/ChangeLog | 1293 +++++++++-
b/addons/ooh323c/src/ooCalls.c | 3
b/addons/ooh323c/src/ooGkClient.c | 1
b/addons/ooh323c/src/oochannels.c | 43
b/addons/ooh323c/src/ooq931.c | 5
b/apps/app_dial.c | 1
b/apps/app_echo.c | 3
b/apps/app_queue.c | 13
b/apps/app_voicemail.c | 2
b/asterisk-13.13.0-rc1-summary.html | 308 ++
b/asterisk-13.13.0-rc1-summary.txt | 764 +++++
b/bridges/bridge_builtin_features.c | 2
b/bridges/bridge_softmix.c | 28
b/channels/chan_pjsip.c | 237 +
b/channels/chan_rtp.c | 2
b/channels/chan_sip.c | 18
b/channels/chan_unistim.c | 11
b/configs/basic-pbx/pjsip.conf | 3
b/configs/samples/asterisk.conf.sample | 9
b/configs/samples/codecs.conf.sample | 54
b/configs/samples/pjsip.conf.sample | 11
b/configs/samples/rtp.conf.sample | 12
b/configure | 220 +
b/configure.ac | 12
b/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py | 31
b/contrib/realtime/mssql/mssql_config.sql | 14
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 14
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/doc/appdocsxml.xslt | 20
b/include/asterisk.h | 9
b/include/asterisk/_private.h | 1
b/include/asterisk/autoconfig.h.in | 6
b/include/asterisk/bridge.h | 9
b/include/asterisk/channel.h | 61
b/include/asterisk/channel_internal.h | 2
b/include/asterisk/file.h | 28
b/include/asterisk/manager.h | 2
b/include/asterisk/module.h | 7
b/include/asterisk/options.h | 2
b/include/asterisk/res_pjsip.h | 2
b/include/asterisk/rtp_engine.h | 3
b/include/asterisk/stasis_app.h | 10
b/include/asterisk/stasis_bridges.h | 4
b/include/asterisk/tcptls.h | 1
b/include/asterisk/vector.h | 8
b/main/Makefile | 11
b/main/asterisk.c | 48
b/main/astobj2.c | 4
b/main/autoservice.c | 66
b/main/bridge.c | 34
b/main/bridge_channel.c | 3
b/main/cdr.c | 19
b/main/channel.c | 160 -
b/main/channel_internal_api.c | 29
b/main/cli.c | 14
b/main/codec_builtin.c | 16
b/main/features_config.c | 2
b/main/file.c | 137 +
b/main/format_cap.c | 2
b/main/loader.c | 5
b/main/manager_bridges.c | 52
b/main/manager_channels.c | 11
b/main/netsock.c | 2
b/main/rtp_engine.c | 87
b/main/stasis_bridges.c | 29
b/main/tcptls.c | 67
b/main/utils.c | 244 +
b/menuselect/aclocal.m4 | 281 ++
b/menuselect/configure | 197 +
b/menuselect/configure.ac | 9
b/res/ari/ari_model_validators.c | 463 +++
b/res/ari/ari_model_validators.h | 65
b/res/ari/ari_websockets.c | 2
b/res/ari/resource_bridges.c | 66
b/res/ari/resource_bridges.h | 28
b/res/ari/resource_channels.c | 7
b/res/res_agi.c | 38
b/res/res_ari_bridges.c | 146 +
b/res/res_ari_channels.c | 2
b/res/res_format_attr_opus.c | 10
b/res/res_http_websocket.c | 19
b/res/res_pjsip.c | 137 -
b/res/res_pjsip/include/res_pjsip_private.h | 14
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_message_ip_updater.c | 303 ++
b/res/res_pjsip_caller_id.c | 14
b/res/res_pjsip_outbound_authenticator_digest.c | 13
b/res/res_pjsip_outbound_registration.c | 2
b/res/res_pjsip_pubsub.c | 20
b/res/res_pjsip_registrar_expire.c | 2
b/res/res_pjsip_sdp_rtp.c | 54
b/res/res_pjsip_session.c | 15
b/res/res_pjsip_t38.c | 13
b/res/res_rtp_asterisk.c | 107
b/res/res_stasis.c | 22
b/res/stasis/app.c | 105
b/res/stasis/app.h | 26
b/res/stasis/cli.c | 216 +
b/res/stasis/cli.h | 43
b/res/stasis_recording/stored.c | 217 -
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 2
b/rest-api/api-docs/bridges.json | 84
b/rest-api/api-docs/channels.json | 10
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 22
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/tests/test_astobj2_thrash.c | 11
b/tests/test_file.c | 197 +
b/tests/test_res_stasis.c | 6
b/third-party/pjproject/Makefile | 75
b/third-party/pjproject/Makefile.rules | 10
b/third-party/pjproject/apply_patches | 4
b/third-party/pjproject/configure.m4 | 5
b/third-party/pjproject/patches/0000-remove-third-party.patch | 142 +
b/third-party/pjproject/patches/0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch | 134 +
b/third-party/pjproject/patches/0006-r5473-svn-backport-Fix-pending-query.patch | 28
b/third-party/pjproject/patches/0006-r5475-svn-backport-Remove-DNS-cache-entry.patch | 70
b/third-party/pjproject/patches/0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch | 33
res/res_pjsip_multihomed.c | 225 -
131 files changed, 7190 insertions(+), 2801 deletions(-)</pre><br></html>

View File

@@ -1,760 +0,0 @@
Release Summary
asterisk-13.13.0
Date: 2016-11-22
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Open Issues
5. Other Changes
6. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-13.12.0.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were affected by commits that went into
this release.
Coders Testers Reporters
20 gtjoseph 1 Dmitry Melekhov 7 Matt Jordan
11 Joshua Colp 5 Alexander Traud
10 Matt Jordan 4 Morten Tryfoss
10 Mark Michelson 4 scgm11
7 Richard Mudgett 4 Joshua Colp
3 Kevin Harwell 3 George Joseph
3 Tzafrir Cohen 3 Richard Mudgett
3 Alexander Traud 2 Gabriele Giacone
3 Corey Farrell <1o5g4r8o@gmail.com>
3 Alexander Anikin 2 Andrew Nagy
2 Sebastian Gutierrez 1 Rusty Newton
1 Michael Walton 1 Dmitry Melekhov
1 Etienne Lessard 1 Andreas Wetzel
1 Leandro Dardini 1 Ian Gilmour
1 snuffy 1 Alexei Gradinari
1 Pascal Cadotte Michaud 1 Bill Brigden
1 Maciej Szmigiero 1 Andrew Nagy
1 Michael Kuron 1 Sergey Grachev
1 Rusty Newton 1 snuffy
1 Grachev Sergey 1 Daniele Pallastrelli
1 Alexei Gradinari 1 Kevin Harwell
1 Igor Goncharovskiy 1 Kayode
1 Moises Silva 1 Michael Keuter
1 Dmitry Melekhov
1 Harley Peters
1 Corey Farrell
1 Leandro Dardini
1 Jonathan Harris
1 Badalian Vyacheslav
1 Doug Lytle
1 scgm11
1 Richard Mudgett
1 Maciej Szmigiero
1 Etienne Lessard
1 Tzafrir Cohen
1 John Kiniston
1 Jason
1 Florian Loyau
1 Ian Gilmour
1 Michael Walton
1 Morton Tryfoss
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
New Feature
Category: General
ASTERISK-26595: ARI: Add the ability to control the source of video in a
multi-party mixing bridge
Reported by: Matt Jordan
* [d23b4af477] Matt Jordan -- res/ari/resource_bridges: Add the ability
to manipulate the video source
ASTERISK-26470: ARI: Add an 'asterisk_id' field to outgoing events
Reported by: Matt Jordan
* [42cfdcd1b7] Matt Jordan -- res/ari: Add the Asterisk EID field to
outgoing events
Bug
Category: Addons/chan_ooh323
ASTERISK-24400: ooh323 sends wrong hangup code
Reported by: Dmitry Melekhov
* [a9ac1f5de4] Alexander Anikin -- chan_ooh323: Fixes to work right with
Cisco devices
Category: Applications/app_dial
ASTERISK-26549: app_dial: When PickupChan() is used some channels may have
incorrect device state
Reported by: Joshua Colp
* [d971647949] Joshua Colp -- app_dial: Fix incorrect device state when
channel is picked up.
Category: Applications/app_queue
ASTERISK-26462: [patch] app_queue: While using queues with realtime,
setting back to an empty context doesn't stop the exit key usage
Reported by: Leandro Dardini
* [0306869399] Leandro Dardini -- app_queue: Added initialization for
"context" parameter
Category: Applications/app_voicemail
ASTERISK-26503: app_voicemail: Asterisk crashes when MailboxExists is used
Reported by: Doug Lytle
* [14496ce1e5] Joshua Colp -- app_voicemail: Clear voice mailbox in
MailboxExists and MAILBOX_EXISTS.
Category: Bridges/bridge_softmix
ASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt Jordan
* [e7dc536b7a] Matt Jordan -- main/bridge_channel: Fix channel reference
leak on video source
* [7c824b955d] Matt Jordan -- main/bridge: Add some verbose logging for
video source changes
* [fd6af2dee8] Matt Jordan -- bridges/bridge_softmix: Remove SSRC
changes on join/leave; update video source
Category: Channels/chan_dahdi
ASTERISK-26412: build: Prepare for gcc 6.2
Reported by: George Joseph
* [bd4d7d8ad0] Kevin Harwell -- stasis_recording/stored: remove calls to
deprecated readdir_r function.
Category: Channels/chan_pjsip
ASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard Mudgett
* [e632222bc4] Richard Mudgett --
res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.
* [afecb2cfc0] Richard Mudgett -- bundled pjproject: Fix DNS write to
freed memory.
ASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John Kiniston
* [c2036c827c] snuffy -- Fix issue with CLI not returning to prompt
after running "features show"
ASTERISK-26482: [patch] chan_pjsip: segfault on already disconnected
session
Reported by: Alexei Gradinari
* [6d462b9eaf] Alexei Gradinari -- chan_pjsip: segfault on already
disconnected session
Category: Channels/chan_sip/General
ASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls
after 2 minutes - rtptimeout behaving badly - regression
Reported by: Michael Keuter
* [cb30963d22] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
updated"
ASTERISK-26476: chan_sip: Incorrect display option "Outbound reg. retry
403" in "sip show settings"
Reported by: Sergey Grachev
* [b3f10b7b94] Grachev Sergey -- chan_sip: Incorrect display option
Outbound reg. retry 403
ASTERISK-26457: [patch] force_rport,auto_comedia: No NAT detection
triggered.
Reported by: Alexander Traud
* [a859bcb49c] Alexander Traud -- chan_sip: Support nat=auto_comedia or
nat=force_rport,auto_comedia.
Category: Channels/chan_unistim
ASTERISK-26565: chan_unistim on 11, 13, 14 placing call on hold
temporarily locks up set
Reported by: Jason
* [3faca1d4ff] Igor Goncharovskiy -- Fix closing rtp ports after call
finished in chan_unistim.
Category: Codecs/codec_opus
ASTERISK-26520: codec_opus: Generated fmtp line has no content
Reported by: scgm11
* [2c031b67d3] Mark Michelson -- res_format_attr_opus: Fix fmtp
generation.
Category: Core/AstMM
ASTERISK-26526: [UBSAN] vector.h: null pointer can be passed as argument 2
to memcpy
Reported by: Badalian Vyacheslav
* [30b1bc77d2] Corey Farrell -- vector: Prevent NULL argument to memcpy.
ASTERISK-26524: astobj2: data_size variable is wasted space when AO2_DEBUG
is not enabled.
Reported by: Corey Farrell
* [b96f18560b] Corey Farrell -- astobj2: Declare private variable
data_size for AO2_DEBUG only.
Category: Core/Bridging
ASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt Jordan
* [e7dc536b7a] Matt Jordan -- main/bridge_channel: Fix channel reference
leak on video source
* [7c824b955d] Matt Jordan -- main/bridge: Add some verbose logging for
video source changes
* [fd6af2dee8] Matt Jordan -- bridges/bridge_softmix: Remove SSRC
changes on join/leave; update video source
Category: Core/BuildSystem
ASTERISK-26608: Compile and link failures on OpenBSD
Reported by: snuffy
* [b213045fe4] gtjoseph -- build: Various OpenBSD issues
ASTERISK-26592: Latest libedit (3.1) defaults to unicode and makes
asterisk CLI read garbage
Reported by: George Joseph
* [5e0c224043] gtjoseph -- cli: Fix ast_el_read_char to work with
libedit >= 3.1
ASTERISK-22480: Embedded pjproject: build.mak contains hardcoded full path
to version.mak
Reported by: Matt Jordan
* [61a5c3460e] gtjoseph -- pjproject_bundled: Remove usage of tar's
--strip-components option
ASTERISK-26356: menuselect: invalid test for GTK2
Reported by: Tzafrir Cohen
* [6f5880913f] Tzafrir Cohen -- menuselect: invalid test for GTK2
Category: Core/CodecInterface
ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but
codec_opus not loaded.
Reported by: Richard Mudgett
* [ed0f1afc8c] Richard Mudgett -- codec_opus: Fix warning when Opus
negotiated but codec_opus not loaded.
Category: Core/General
ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but
codec_opus not loaded.
Reported by: Richard Mudgett
* [ed0f1afc8c] Richard Mudgett -- codec_opus: Fix warning when Opus
negotiated but codec_opus not loaded.
ASTERISK-26509: A few non-critical deprecation warnings when building on
Ubuntu 16.10
Reported by: Jonathan Harris
* [bd4d7d8ad0] Kevin Harwell -- stasis_recording/stored: remove calls to
deprecated readdir_r function.
Category: Core/ManagerInterface
ASTERISK-26537: AMI: NewConnectedLine event is not documented
Reported by: Etienne Lessard
* [42bd70b29f] Etienne Lessard -- manager: Add documentation for
NewConnectedLine event.
Category: Core/RTP
ASTERISK-26311: [patch] rtp_engine: Allow more than 32 dynamic payload
types.
Reported by: Alexander Traud
* [0cf1778eed] Alexander Traud -- rtp_engine: Allow more than 32 dynamic
payload types.
Category: Core/Stasis
ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI
calls
Reported by: Daniele Pallastrelli
* [3a1f9c5dab] Joshua Colp -- res_stasis: Don't unsubscribe from a NULL
bridge.
Category: Documentation
ASTERISK-26514: Super Awesome Company: Don't specify transport in
pjsip.conf
Reported by: Rusty Newton
* [87903a6848] Rusty Newton -- SAC documentation: don't specify
transports for endpoints and registrations
Category: Features
ASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John Kiniston
* [c2036c827c] snuffy -- Fix issue with CLI not returning to prompt
after running "features show"
Category: General
ASTERISK-26575: testsuite: Need to check PJSIP functionality when res_srtp
is not loaded.
Reported by: Joshua Colp
* [b70eb07c53] Joshua Colp -- res_pjsip_sdp_rtp: Reject offer of
required SRTP without res_srtp.
ASTERISK-25070: Fix FTBFS on Hurd
Reported by: Gabriele Giacone
* [94c9496ed5] Tzafrir Cohen -- netsock.c: fix includes for HURD
* [c1c9487375] Tzafrir Cohen -- define PATH_MAX for HURD
ASTERISK-26387: Asterisk segfaults shortly after starting even with no
active calls.
Reported by: Harley Peters
* [d7f457e4c1] Richard Mudgett -- bundled pjproject: Crashes while
resolving DNS names.
ASTERISK-26513: tests/channels/pjsip/qualify/auth: Crashing enough to be a
nuisance
Reported by: Joshua Colp
* [f373de3020] Corey Farrell -- Fix shutdown crash caused by modules
being left open.
ASTERISK-26421: Segmentation Fault with ARI originate into mixing bridge
with 43 clients
Reported by: Andrew Nagy
* [eff97808fb] Mark Michelson -- ARI: Detect duplicate channel IDs
* [012fda29d2] Mark Michelson -- CDR: Alter destruction pattern for CDR
chains.
ASTERISK-26480: [patch] CLI: core set debug: Auto-completes File not
Module
Reported by: Alexander Traud
* [662b560c35] Alexander Traud -- cli: Auto-complete File not Module for
core set debug.
Category: Resources/res_agi
ASTERISK-26343: ASTERISK-25951 causes issues for callerid manipulation
through agi
Reported by: Morten Tryfoss
* [404596b790] gtjoseph -- channel: Fix issues in hangup scenarios
caused by frame deferral
* [2e3a354575] Mark Michelson -- autoservice: Use frame deferral API
* [5c10091f3d] Mark Michelson -- AGI: Only defer frames when in an
interception routine.
* [9231a56cf3] Mark Michelson -- Add API for channel frame deferral.
Category: Resources/res_ari_bridges
ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI
calls
Reported by: Daniele Pallastrelli
* [3a1f9c5dab] Joshua Colp -- res_stasis: Don't unsubscribe from a NULL
bridge.
Category: Resources/res_pjsip
ASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard Mudgett
* [e632222bc4] Richard Mudgett --
res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.
* [afecb2cfc0] Richard Mudgett -- bundled pjproject: Fix DNS write to
freed memory.
Category: Resources/res_pjsip_caller_id
ASTERISK-26307: res_pjsip_caller_id: Crash on outgoing change
Reported by: Bill Brigden
* [675c71ae8c] Joshua Colp -- res_pjsip_caller_id: Fix crash on session
timers UPDATE on inbound calls.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-26541: res_pjsip_sdp_rtp: Restrict number of formats to maximum
Reported by: Joshua Colp
* [5f188bb7a8] Joshua Colp -- res_pjsip_sdp_rtp: Limit number of formats
to defined maximum.
ASTERISK-26423: res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio
loss and wonkiness
Reported by: Andreas Wetzel
* [e0bc17edff] Joshua Colp -- pjsip: Fix a few media bugs with reinvites
and asymmetric payloads.
ASTERISK-26309: [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack)
installations.
Reported by: Alexander Traud
* [f534f67f52] Joshua Colp -- res_pjsip_sdp_rtp: Fix address family of
explicit media_address.
* [bb982480d8] Joshua Colp -- pjsip: Support dual stack automatically.
Category: Third-Party/pjproject
ASTERISK-26510: pjproject_bundled uses the --strip-components option of
tar which isn't supported in older versions
Reported by: George Joseph
* [61a5c3460e] gtjoseph -- pjproject_bundled: Remove usage of tar's
--strip-components option
Category: pjproject/pjsip
ASTERISK-26344: Asterisk 13.11.0 + PJSIP crash
Reported by: Ian Gilmour
* [d7f457e4c1] Richard Mudgett -- bundled pjproject: Crashes while
resolving DNS names.
Improvement
Category: Applications/app_queue
ASTERISK-26558: app_queue: add variable to know if the call is not
answered after a queue
Reported by: scgm11
* [71dc333565] Joshua Colp -- app_queue: Add mention of 'ABANDON'
variable to CHANGES.
* [7fd5031c1c] Sebastian Gutierrez -- app_queue: new variable set when
abandoned
Category: Channels/chan_sip/General
ASTERISK-26176: chan_sip: Add AccountCode to AMI PeerEntry
Reported by: scgm11
* [714412f6c4] Sebastian Gutierrez -- chan_sip: add missing account code
Category: Codecs/codec_opus
ASTERISK-26538: codec_opus: Add sample to
configs/samples/codecs.conf.sample
Reported by: Kevin Harwell
* [50fa868ab8] Kevin Harwell -- codecs.conf.sample: Add sample and
option descriptions for codec_opus
Category: Core/General
ASTERISK-25063: [patch]add X.509 subject alternative name support to
Asterisk TLS support
Reported by: Maciej Szmigiero
* [7b96e8cc3d] Maciej Szmigiero -- Add X.509 subject alternative name
support to TLS certificate
Category: Resources/res_ari
ASTERISK-26488: ARI: Add 'ari show app', 'ari show apps', and 'ari set
debug' CLI commands
Reported by: Matt Jordan
* [29692d4aa4] Matt Jordan -- res/stasis: Add CLI commands for
displaying/debugging ARI apps
Category: Resources/res_rtp_asterisk
ASTERISK-26418: res_rtp_asterisk: Speed up ICE resolution by blacklisting
host subnets that are not involved in RTP
Reported by: Michael Walton
* [3c62b60e56] Michael Walton -- res_rtp_asterisk: Add ice_blacklist
option
----------------------------------------------------------------------
Open Issues
[Back to Top]
This is a list of all open issues from the issue tracker that were
referenced by changes that went into this release.
Bug
Category: Channels/chan_multicast_rtp
ASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode
* [644fad7477] Moises Silva -- chan_rtp: Set a sane default rtp engine
for unicast.
Category: Core/Jitterbuffer
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using
JitterBuffers of any kind
Reported by: Florian Loyau
* [cb30963d22] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
updated"
Category: Core/RTP
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using
JitterBuffers of any kind
Reported by: Florian Loyau
* [cb30963d22] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always
updated"
Category: Resources/res_rtp_multicast
ASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode
* [644fad7477] Moises Silva -- chan_rtp: Set a sane default rtp engine
for unicast.
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+-----------------+-----------------------------------------|
| 751d43e8e4 | Joshua Colp | Update for 13.13.0-rc1 |
|------------+-----------------+-----------------------------------------|
| cb624b10ae | Mark Michelson | Bump ARI version to 1.10.0 |
|------------+-----------------+-----------------------------------------|
| bde3d022a3 | Mark Michelson | manager: update minor version |
|------------+-----------------+-----------------------------------------|
| c92dcc76da | gtjoseph | file.c/__ast_file_read_dirs: Fix issues |
| | | on filesystems without d_type |
|------------+-----------------+-----------------------------------------|
| 0790aa528a | Matt Jordan | pjproject: Use a much higher limit for |
| | | PJ_ICE_MAX_CHECKS |
|------------+-----------------+-----------------------------------------|
| 993a6f96c7 | Matt Jordan | apps/app_echo: Only relay a single |
| | | video source change frame |
|------------+-----------------+-----------------------------------------|
| 404a62eeee | gtjoseph | Revert "Revert "channel: Use frame |
| | | deferral API for safe sleep."" |
|------------+-----------------+-----------------------------------------|
| 09d8febc91 | gtjoseph | Revert "Revert "autoservice: Use frame |
| | | deferral API"" |
|------------+-----------------+-----------------------------------------|
| ffad2b44df | gtjoseph | Revert "Revert "AGI: Only defer frames |
| | | when in an interception routine."" |
|------------+-----------------+-----------------------------------------|
| 2fefb6187f | gtjoseph | Revert "Revert "Add API for channel |
| | | frame deferral."" |
|------------+-----------------+-----------------------------------------|
| 412d43fa21 | Richard Mudgett | res_pjsip.c: Rework |
| | | endpt_send_request() req_wrapper code. |
|------------+-----------------+-----------------------------------------|
| 2e7fc56d3c | Richard Mudgett | res_pjsip: Fix tdata leaks in off |
| | | nominal paths. |
|------------+-----------------+-----------------------------------------|
| da68b185b3 | Richard Mudgett | res_pjsip_registrar_expire.c: Remove |
| | | extra linefeed in debug message. |
|------------+-----------------+-----------------------------------------|
| 6b5a7ced13 | gtjoseph | Revert "Add API for channel frame |
| | | deferral." |
|------------+-----------------+-----------------------------------------|
| 6be5d8de0d | gtjoseph | Revert "AGI: Only defer frames when in |
| | | an interception routine." |
|------------+-----------------+-----------------------------------------|
| 1df434e2b4 | gtjoseph | Revert "autoservice: Use frame deferral |
| | | API" |
|------------+-----------------+-----------------------------------------|
| 58c88cfbaa | gtjoseph | Revert "channel: Use frame deferral API |
| | | for safe sleep." |
|------------+-----------------+-----------------------------------------|
| a562fbe618 | gtjoseph | build: Fix default values for some |
| | | SANITIZER options |
|------------+-----------------+-----------------------------------------|
| e043d1a55c | Mark Michelson | res_pjsip_session: Do not call session |
| | | supplements when it's too late. |
|------------+-----------------+-----------------------------------------|
| 44f7e25239 | Mark Michelson | channel: Use frame deferral API for |
| | | safe sleep. |
|------------+-----------------+-----------------------------------------|
| 0ee249075a | Alexander | chan_ooh323: reset rrq count on gk |
| | Anikin | registration |
|------------+-----------------+-----------------------------------------|
| 59c23e1768 | Michael Kuron | automon: restore mixing of the both |
| | | channels after recording stops |
|------------+-----------------+-----------------------------------------|
| e79acaeb75 | Matt Jordan | res_http_websocket: Increase the buffer |
| | | size for non-LOW_MEMORY systems |
|------------+-----------------+-----------------------------------------|
| 7a83196985 | Matt Jordan | res_stasis: Set a video source mode on |
| | | Stasis created bridges |
|------------+-----------------+-----------------------------------------|
| eceab15f33 | Alexander | chan_ooh323: Fix infinite loop on read |
| | Anikin | second part of H.225 packet |
|------------+-----------------+-----------------------------------------|
| a9992da4aa | gtjoseph | pjproject_bundled: Fix issue with |
| | | libasteriskpj needing libresample |
|------------+-----------------+-----------------------------------------|
| a36a7d0cf4 | gtjoseph | pjproject_bundled: Fix compile of pjsua |
| | | so it handles audio |
|------------+-----------------+-----------------------------------------|
| 6b1c55dc9b | gtjoseph | pjproject_bundled: Fix issue where |
| | | "/version.mak" wasn't found |
|------------+-----------------+-----------------------------------------|
| 3a2092b722 | gtjoseph | test_astobj2_thrash: Fix multithreaded |
| | | issues |
|------------+-----------------+-----------------------------------------|
| 640203802e | Pascal Cadotte | typo: s/paranthesis/parenthesis/ in a |
| | Michaud | comment |
|------------+-----------------+-----------------------------------------|
| 9b3557e054 | gtjoseph | pjproject_bundled: Fixed various build |
| | | issues |
|------------+-----------------+-----------------------------------------|
| 74d9385273 | gtjoseph | utils.c: Fix ast_set_default_eid for |
| | | multiple platforms |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
asterisk-13.12.0-summary.html | 543 ----
asterisk-13.12.0-summary.txt | 1275 ---------
b/.version | 2
b/CHANGES | 95
b/ChangeLog | 1293 +++++++++-
b/addons/ooh323c/src/ooCalls.c | 3
b/addons/ooh323c/src/ooGkClient.c | 1
b/addons/ooh323c/src/oochannels.c | 43
b/addons/ooh323c/src/ooq931.c | 5
b/apps/app_dial.c | 1
b/apps/app_echo.c | 3
b/apps/app_queue.c | 13
b/apps/app_voicemail.c | 2
b/asterisk-13.13.0-rc1-summary.html | 308 ++
b/asterisk-13.13.0-rc1-summary.txt | 764 +++++
b/bridges/bridge_builtin_features.c | 2
b/bridges/bridge_softmix.c | 28
b/channels/chan_pjsip.c | 237 +
b/channels/chan_rtp.c | 2
b/channels/chan_sip.c | 18
b/channels/chan_unistim.c | 11
b/configs/basic-pbx/pjsip.conf | 3
b/configs/samples/asterisk.conf.sample | 9
b/configs/samples/codecs.conf.sample | 54
b/configs/samples/pjsip.conf.sample | 11
b/configs/samples/rtp.conf.sample | 12
b/configure | 220 +
b/configure.ac | 12
b/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py | 31
b/contrib/realtime/mssql/mssql_config.sql | 14
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 14
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/doc/appdocsxml.xslt | 20
b/include/asterisk.h | 9
b/include/asterisk/_private.h | 1
b/include/asterisk/autoconfig.h.in | 6
b/include/asterisk/bridge.h | 9
b/include/asterisk/channel.h | 61
b/include/asterisk/channel_internal.h | 2
b/include/asterisk/file.h | 28
b/include/asterisk/manager.h | 2
b/include/asterisk/module.h | 7
b/include/asterisk/options.h | 2
b/include/asterisk/res_pjsip.h | 2
b/include/asterisk/rtp_engine.h | 3
b/include/asterisk/stasis_app.h | 10
b/include/asterisk/stasis_bridges.h | 4
b/include/asterisk/tcptls.h | 1
b/include/asterisk/vector.h | 8
b/main/Makefile | 11
b/main/asterisk.c | 48
b/main/astobj2.c | 4
b/main/autoservice.c | 66
b/main/bridge.c | 34
b/main/bridge_channel.c | 3
b/main/cdr.c | 19
b/main/channel.c | 160 -
b/main/channel_internal_api.c | 29
b/main/cli.c | 14
b/main/codec_builtin.c | 16
b/main/features_config.c | 2
b/main/file.c | 137 +
b/main/format_cap.c | 2
b/main/loader.c | 5
b/main/manager_bridges.c | 52
b/main/manager_channels.c | 11
b/main/netsock.c | 2
b/main/rtp_engine.c | 87
b/main/stasis_bridges.c | 29
b/main/tcptls.c | 67
b/main/utils.c | 244 +
b/menuselect/aclocal.m4 | 281 ++
b/menuselect/configure | 197 +
b/menuselect/configure.ac | 9
b/res/ari/ari_model_validators.c | 463 +++
b/res/ari/ari_model_validators.h | 65
b/res/ari/ari_websockets.c | 2
b/res/ari/resource_bridges.c | 66
b/res/ari/resource_bridges.h | 28
b/res/ari/resource_channels.c | 7
b/res/res_agi.c | 38
b/res/res_ari_bridges.c | 146 +
b/res/res_ari_channels.c | 2
b/res/res_format_attr_opus.c | 10
b/res/res_http_websocket.c | 19
b/res/res_pjsip.c | 137 -
b/res/res_pjsip/include/res_pjsip_private.h | 14
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_message_ip_updater.c | 303 ++
b/res/res_pjsip_caller_id.c | 14
b/res/res_pjsip_outbound_authenticator_digest.c | 13
b/res/res_pjsip_outbound_registration.c | 2
b/res/res_pjsip_pubsub.c | 20
b/res/res_pjsip_registrar_expire.c | 2
b/res/res_pjsip_sdp_rtp.c | 54
b/res/res_pjsip_session.c | 15
b/res/res_pjsip_t38.c | 13
b/res/res_rtp_asterisk.c | 107
b/res/res_stasis.c | 22
b/res/stasis/app.c | 105
b/res/stasis/app.h | 26
b/res/stasis/cli.c | 216 +
b/res/stasis/cli.h | 43
b/res/stasis_recording/stored.c | 217 -
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 2
b/rest-api/api-docs/bridges.json | 84
b/rest-api/api-docs/channels.json | 10
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 22
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/tests/test_astobj2_thrash.c | 11
b/tests/test_file.c | 197 +
b/tests/test_res_stasis.c | 6
b/third-party/pjproject/Makefile | 75
b/third-party/pjproject/Makefile.rules | 10
b/third-party/pjproject/apply_patches | 4
b/third-party/pjproject/configure.m4 | 5
b/third-party/pjproject/patches/0000-remove-third-party.patch | 142 +
b/third-party/pjproject/patches/0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch | 134 +
b/third-party/pjproject/patches/0006-r5473-svn-backport-Fix-pending-query.patch | 28
b/third-party/pjproject/patches/0006-r5475-svn-backport-Remove-DNS-cache-entry.patch | 70
b/third-party/pjproject/patches/0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch | 33
res/res_pjsip_multihomed.c | 225 -
131 files changed, 7190 insertions(+), 2801 deletions(-)