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Author SHA1 Message Date
kharwell
56487be6d9 ChangeLog: Updated for 13.6.0 2015-10-09 16:48:48 -05:00
kharwell
68d2a14e1a Release summaries: Add summaries for 13.6.0 2015-10-09 16:48:41 -05:00
Kevin Harwell
cc0eff5651 Release summaries: Remove previous versions 2015-10-09 16:45:28 -05:00
kharwell
8cd191b885 .version: Update for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
a3777c24fd .lastclean: Update for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
68121cef21 realtime: Add database scripts for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
d72dab4f40 ChangeLog: Updated for 13.6.0-rc3 2015-10-07 13:42:02 -05:00
kharwell
9da83dbd15 Release summaries: Add summaries for 13.6.0-rc3 2015-10-07 13:41:55 -05:00
Kevin Harwell
8c60f9326c Release summaries: Remove previous versions 2015-10-07 13:41:47 -05:00
kharwell
316d47755b .version: Update for 13.6.0-rc3 2015-10-07 13:41:47 -05:00
kharwell
74a86d0a72 .lastclean: Update for 13.6.0-rc3 2015-10-07 13:41:47 -05:00
kharwell
4c39bea6f0 realtime: Add database scripts for 13.6.0-rc3 2015-10-07 13:41:46 -05:00
Matt Jordan
10e790f81a res/res_rtp_asterisk: Fix assignment after ao2 decrement
When we decide we will no longer schedule an RTCP write, we remove the
reference to the RTP instance, then assign -1 to the stored scheduler ID
in case something else comes along and wants to see if anything is scheduled.

That scheduler ID is on the RTP instance. After 60a9172d7e was merged to
fix the regression introduced by 3cf0f29310, this improper assignment on a
potentially destroyed object started getting tripped on the build agents.

Frankly, this should have been crashing a lot more often earlier. I can only
assume that the timing was changed just enough by both changes to start
actually hitting this problem.

As it is, simply moving the assignment prior to the ao2 deference is sufficient
to keep the RTP instance from being referenced when it is very, truly,
aboslutely dead.

(Note that it is still good practice to assign -1 to the scheduler ID when we
know we won't be scheduling it again, as the ao2 deref *may* not always destroy
the ao2 object.)

ASTERISK-25449

Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
2015-10-06 20:52:21 -05:00
7 changed files with 1239 additions and 150 deletions

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13.6.0-rc2
13.6.0

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2015-10-09 21:48 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.6.0 Released.
2015-10-09 16:48 +0000 [68d2a14e1a] Kevin Harwell <kharwell@lunkwill>
* Release summaries: Add summaries for 13.6.0
2015-10-09 16:45 +0000 [cc0eff5651] Kevin Harwell <kharwell@lunkwill.digium.internal>
* Release summaries: Remove previous versions
2015-10-09 16:45 +0000 [8cd191b885] Kevin Harwell <kharwell@lunkwill>
* .version: Update for 13.6.0
2015-10-09 16:45 +0000 [a3777c24fd] Kevin Harwell <kharwell@lunkwill>
* .lastclean: Update for 13.6.0
2015-10-09 16:45 +0000 [68121cef21] Kevin Harwell <kharwell@lunkwill>
* realtime: Add database scripts for 13.6.0
2015-10-07 18:42 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.6.0-rc3 Released.
2015-10-07 13:41 +0000 [9da83dbd15] Kevin Harwell <kharwell@lunkwill>
* Release summaries: Add summaries for 13.6.0-rc3
2015-10-07 13:41 +0000 [8c60f9326c] Kevin Harwell <kharwell@lunkwill.digium.internal>
* Release summaries: Remove previous versions
2015-10-07 13:41 +0000 [316d47755b] Kevin Harwell <kharwell@lunkwill>
* .version: Update for 13.6.0-rc3
2015-10-07 13:41 +0000 [74a86d0a72] Kevin Harwell <kharwell@lunkwill>
* .lastclean: Update for 13.6.0-rc3
2015-10-07 13:41 +0000 [4c39bea6f0] Kevin Harwell <kharwell@lunkwill>
* realtime: Add database scripts for 13.6.0-rc3
2015-10-06 20:43 +0000 [10e790f81a] Matt Jordan <mjordan@digium.com>
* res/res_rtp_asterisk: Fix assignment after ao2 decrement
When we decide we will no longer schedule an RTCP write, we remove the
reference to the RTP instance, then assign -1 to the stored scheduler ID
in case something else comes along and wants to see if anything is scheduled.
That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to
fix the regression introduced by 3cf0f29310, this improper assignment on a
potentially destroyed object started getting tripped on the build agents.
Frankly, this should have been crashing a lot more often earlier. I can only
assume that the timing was changed just enough by both changes to start
actually hitting this problem.
As it is, simply moving the assignment prior to the ao2 deference is sufficient
to keep the RTP instance from being referenced when it is very, truly,
aboslutely dead.
(Note that it is still good practice to assign -1 to the scheduler ID when we
know we won't be scheduling it again, as the ao2 deref *may* not always destroy
the ao2 object.)
ASTERISK-25449
Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
2015-10-06 18:40 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.6.0-rc2 Released.

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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-13.6.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.6.0-rc2</h3><h3 align="center">Date: 2015-10-06</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="#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-13.6.0-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%">3 Kevin Harwell <kharwell@lunkwill><br/>1 Matt Jordan <mjordan@digium.com><br/>1 Kevin Harwell <kharwell@lunkwill.digium.internal><br/>1 Joshua Colp <jcolp@digium.com><br/></td><td width="33%"><td width="33%">1 Joshua Colp <jcolp@digium.com><br/>1 Matt Jordan <mjordan@digium.com><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>Bug</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25449">ASTERISK-25449</a>: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=754daeca0a48e5c9365cc7fa4c5a3da6c61ae7f6">[754daeca0a]</a> Matt Jordan -- Fix improper usage of scheduler exposed by 5c713fdf18f</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25438">ASTERISK-25438</a>: res_rtp_asterisk: ICE role message even when ICE is not enabled<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9913d47697716040917b49c0c60ae2d98493d516">[9913d47697]</a> Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug.</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=dd74af7e460083df8ca774db2793791ff6413c6e">dd74af7e46</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a11a78ca34fae7ef2069d0a9ae6a5cf1fd2856d7">a11a78ca34</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=570329ec8a05c27675721734e2da3c42c707f0f7">570329ec8a</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51c9ff47f6bab1aed30f95a8601e0ddf253b657e">51c9ff47f6</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc2</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-13.6.0-rc1-summary.html | 298 --------------
asterisk-13.6.0-rc1-summary.txt | 777 ---------------------------------------
b/.version | 2
b/channels/chan_sip.c | 4
b/channels/chan_skinny.c | 6
5 files changed, 6 insertions(+), 1081 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.6.0-rc2
Date: 2015-10-06
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. 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-13.6.0-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
3 Kevin Harwell 1 Joshua Colp
1 Matt Jordan 1 Matt Jordan
1 Kevin Harwell
1 Joshua Colp
----------------------------------------------------------------------
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.
Bug
Category: Core/General
ASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes
erroneous duplicate RTCP messages; other potential scheduling issues in
chan_sip/chan_skinny
Reported by: Matt Jordan
* [754daeca0a] Matt Jordan -- Fix improper usage of scheduler exposed by
5c713fdf18f
Category: Resources/res_rtp_asterisk
ASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not
enabled
Reported by: Joshua Colp
* [9913d47697] Joshua Colp -- res_rtp_asterisk: Move "Set role" warning
to be debug.
----------------------------------------------------------------------
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 |
|------------+---------------+-------------------------------------------|
| dd74af7e46 | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+---------------+-------------------------------------------|
| a11a78ca34 | Kevin Harwell | .version: Update for 13.6.0-rc2 |
|------------+---------------+-------------------------------------------|
| 570329ec8a | Kevin Harwell | .lastclean: Update for 13.6.0-rc2 |
|------------+---------------+-------------------------------------------|
| 51c9ff47f6 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc2 |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-13.6.0-rc1-summary.html | 298 --------------
asterisk-13.6.0-rc1-summary.txt | 777 ---------------------------------------
b/.version | 2
b/channels/chan_sip.c | 4
b/channels/chan_skinny.c | 6
5 files changed, 6 insertions(+), 1081 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-13.6.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.6.0</h3><h3 align="center">Date: 2015-10-09</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-13.5.0.</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%">40 Richard Mudgett <rmudgett@digium.com><br/>18 Kevin Harwell <kharwell@lunkwill><br/>16 Joshua Colp <jcolp@digium.com><br/>15 Mark Michelson <mmichelson@digium.com><br/>13 Matt Jordan <mjordan@digium.com><br/>9 Scott Griepentrog <scott@griepentrog.com><br/>8 Kevin Harwell <kharwell@lunkwill.digium.internal><br/>3 Scott Emidy <jemidy@digium.com><br/>3 David M. Lee <dlee@respoke.io><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Alexander Anikin <may213@yandex.ru><br/>2 Jonathan Rose <jrose@digium.com><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>1 Mark Duncan <mark@syon.co.jp><br/>1 Benjamin Ford <bford@digium.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/></td><td width="33%">1 Elazar Broad<br/></td><td width="33%">13 Matt Jordan <mjordan@digium.com><br/>11 Joshua Colp <jcolp@digium.com><br/>11 Richard Mudgett <rmudgett@digium.com><br/>10 Scott Griepentrog <sgriepentrog@digium.com><br/>8 Mark Michelson<br/>8 Mark Michelson <mmichelson@digium.com><br/>7 John Hardin<br/>5 Kevin Harwell <kharwell@digium.com><br/>4 Scott Griepentrog<br/>3 Richard Mudgett<br/>2 Kevin Harwell<br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Alexandr Dranchuk <alex.dranchuk@gmail.com><br/>2 Dmitriy Serov <serov.d.p@gmail.com><br/>2 Stefan Engström <stefanen@kth.se><br/>1 Oleg Kozlov <olkeep@gmail.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Etienne Lessard<br/>1 Lorne Gaetz<br/>1 Chet Stevens <cwstevens@interact.ccsd.net><br/>1 Rodrigo Ramirez Norambuena <decipher.hk@gmail.com><br/>1 Ashley Sanders<br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Chet Stevens<br/>1 Kevin Scott Adams <ksatllc@att.net><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Sean Pimental<br/>1 Jonathan Rose <jrose@digium.com><br/>1 Etienne Lessard <elessard@avencall.com><br/>1 Alexandr Dranchuk<br/>1 yaron nahum <nachum.yaron@gmail.com><br/>1 Elazar Broad<br/>1 yaron nahum<br/>1 Guido Falsi<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: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25252">ASTERISK-25252</a>: ARI: Add the ability to manipulate log channels<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df9ce3636695781be6ab2479f90766a56747dbd7">[df9ce36366]</a> Scott Emidy -- ARI: Retrieve existing log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e9f1bc08cbda7759707c30b8883b266555d0fefc">[e9f1bc08cb]</a> Scott Emidy -- ARI: Creating log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78364132ce94d9ded24ae6e6ab44b97d256b506d">[78364132ce]</a> Scott Emidy -- ARI: Deleting log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ae762634c317fbcbd98a8c34d2474f7d4b654ed">[1ae762634c]</a> Benjamin Ford -- ARI: Rotate log channels.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25377">ASTERISK-25377</a>: res_pjsip: Change default "From user" from UUID to something more palatable<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac62928d6b7333a0d502be2eba99c238549ae1a3">[ac62928d6b]</a> Mark Michelson -- res_pjsip: Change default from user value.</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25227">ASTERISK-25227</a>: No audio at in-band announcements in ooh323 channel<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71408df2b82c932329d250a6077475e4f51a2b0d">[71408df2b8]</a> Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE with inband info available</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25322">ASTERISK-25322</a>: Crash occurs when using MixMonitor with t() or r() options.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3a56bee83c01dbb538620a11947d420b17cf458">[b3a56bee83]</a> Richard Mudgett -- audiohook.c: Fix MixMonitor crash when using the r() or t() options.</li>
</ul><br><h4>Category: Applications/app_page</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25384">ASTERISK-25384</a>: Regular Asterisk crashes when using Page application. "user_data is NULL"<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f15cd93f0b2cb622d54061515b815e3ebbe76b1">[5f15cd93f0]</a> Richard Mudgett -- app_page.c: Fix crash when forwarding with a predial handler.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25399">ASTERISK-25399</a>: app_queue: AgentComplete event has wrong reason<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4fb95bbc4e4928dd3403a20d401c285a568f0d09">[4fb95bbc4e]</a> Kevin Harwell -- app_queue: AgentComplete event has wrong reason</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25185">ASTERISK-25185</a>: Segfault in app_queue on transfer scenarios<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6409e7b11a2310196a9978b30a6b79e2760be592">[6409e7b11a]</a> Kevin Harwell -- app_queue: Crash when transferring</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25215">ASTERISK-25215</a>: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember<br/>Reported by: Lorne Gaetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5f5b9f384eef0389a7e35d40c91d3586869a125">[e5f5b9f384]</a> Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.</li>
</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25410">ASTERISK-25410</a>: app_record: RECORDED_FILE variable not being populated<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeddee39fb492ea5ec873bdf02ea3858c5282601">[aeddee39fb]</a> Kevin Harwell -- app_record: RECORDED_FILE variable not being populated</li>
</ul><br><h4>Category: Bridges/bridge_holding</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25271">ASTERISK-25271</a>: Parking & blind transfer: Transferer channel not hung up if no MOH<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8458b8d441c2f4143ff135163ff3da4f88fe14c8">[8458b8d441]</a> Jonathan Rose -- holding_bridge: ensure moh participants get frames</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25315">ASTERISK-25315</a>: DAHDI channels send shortened duration DTMF tones.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=256bc52b6684141b100c5018dbf0ad3ce6111585">[256bc52b66]</a> Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=800e0ea48de88c016e1f477edf7db7b9aadc4b54">[800e0ea48d]</a> Richard Mudgett -- chan_dahdi.c: Lock private struct for ast_write().</li>
</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25309">ASTERISK-25309</a>: [patch] iLBC 20 advertised<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f68c995bc97c9b6cb4887043b344087d82aeef10">[f68c995bc9]</a> Alexander Traud -- chan_sip: Fix negotiation of iLBC 30.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25346">ASTERISK-25346</a>: chan_sip: Overwriting answered elsewhere hangup cause on call pickup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c01111223f9dbd383a4dd1cf786b63eff214f238">[c01111223f]</a> Joshua Colp -- chan_sip: Allow call pickup to set the hangup cause.</li>
</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25396">ASTERISK-25396</a>: chan_sip: Extremely long callerid name causes invalid SIP<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b59c4d82b58a7a10e1791f9ed5759af8ac637df2">[b59c4d82b5]</a> Walter Doekes -- chan_sip: Fix From header truncation for extremely long CALLERID(name).</li>
</ul><br><h4>Category: Channels/chan_sip/Security Framework</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25320">ASTERISK-25320</a>: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25af2d71c863062868b8bda6cf83515a1935d27e">[25af2d71c8]</a> Kevin Harwell -- chan_sip.c: wrong peer searched in sip_report_security_event</li>
</ul><br><h4>Category: Channels/chan_skinny</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25353">ASTERISK-25353</a>: [patch] Transcoding while different in Frame size = Frames lost<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b88c54fa4bd537bde46519abb95e30a5f96673ac">[b88c54fa4b]</a> Alexander Traud -- translate: Fix transcoding while different in frame size.</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25341">ASTERISK-25341</a>: bridge: Hangups may get lost when executing actions<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c2dab1e888e59cb429ed61219815bd00eee66c0">[6c2dab1e88]</a> Joshua Colp -- bridge: Kick channel from bridge if hung up during action.</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25449">ASTERISK-25449</a>: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=10e790f81ad3be322193cf948e0334f9af09b00f">[10e790f81a]</a> Matt Jordan -- res/res_rtp_asterisk: Fix assignment after ao2 decrement</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=754daeca0a48e5c9365cc7fa4c5a3da6c61ae7f6">[754daeca0a]</a> Matt Jordan -- Fix improper usage of scheduler exposed by 5c713fdf18f</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25418">ASTERISK-25418</a>: On-hold channels redirected out of a bridge appear to still be on hold<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=629458d34930e5aca56f749bc05562baf95d13f7">[629458d349]</a> Mark Michelson -- Do not swallow frames on channels leaving bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25355">ASTERISK-25355</a>: sched: ast_sched_del may return prematurely due to spurious wakeup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85e1cb51b21c2d194647e16b81b5a1344d2ff911">[85e1cb51b2]</a> Joshua Colp -- sched: ast_sched_del may return prematurely due to spurious wakeup</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25305">ASTERISK-25305</a>: Dynamic logger channels can be added multiple times<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f050fa76eb8535a2b8a3b047527a42ea369d8792">[f050fa76eb]</a> Mark Michelson -- logger: Prevent duplicate dynamic channels from being added.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25407">ASTERISK-25407</a>: Asterisk fails to log to multiple syslog destinations<br/>Reported by: Elazar Broad<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec514ad64dbc0014525008977c8c74c2856c9d3a">[ec514ad64d]</a> Elazar Broad -- core/logging: Fix logging to more than one syslog channel</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25394">ASTERISK-25394</a>: pbx: Incorrect device and presence state when changing hint details<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2bd27d12223fe33b58c453965ed5c6ed3af7c4f5">[2bd27d1222]</a> Joshua Colp -- pbx: Update device and presence state when changing a hint extension.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25367">ASTERISK-25367</a>: pbx: Long pattern match hints may cause "core show hints" to crash<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc1363209e7a5b6c67bf96c593e3beb0884c1fb0">[cc1363209e]</a> Joshua Colp -- pbx: Fix crash when issuing "core show hints" with long pattern match.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25362">ASTERISK-25362</a>: Deadlock due to presence state callback<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03fe79f29eae42be24589b323a5ef3fa9259158d">[03fe79f29e]</a> Mark Michelson -- Fix deadlock on presence state changes.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25325">ASTERISK-25325</a>: ARI PUT reload chan_sip HTTP response 404<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=865377fc38134234f17def6634c47a989cf0e77a">[865377fc38]</a> Rodrigo Ramírez Norambuena -- chan_sip.c: Validation on module reload</li>
</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25312">ASTERISK-25312</a>: res_http_websocket: Terminate connection on fatal cases<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4e9416138339274176cb87b26db905723e553ba">[b4e9416138]</a> Joshua Colp -- res_http_websocket: Forcefully terminate on write errors.</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25369">ASTERISK-25369</a>: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fbf720db91ae8942e9c2ba092179ab2352d44b06">[fbf720db91]</a> Jonathan Rose -- ParkAndAnnounce: Add variable inheritance</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25295">ASTERISK-25295</a>: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5469caa9ddf002d2e75b5fe5dec0c4dbebea1d1e">[5469caa9dd]</a> Joshua Colp -- res_pjsip: Use hash for contact object identity instead of Contact URI.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a676ba2aad5525926ae31b8317b95ae52cbbabbb">[a676ba2aad]</a> Joshua Colp -- taskprocessor: Fix race condition between unreferencing and finding.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25381">ASTERISK-25381</a>: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3e6debdb95a5895894ed2b58b600fcdf17927b9">[c3e6debdb9]</a> Matt Jordan -- res/res_pjsip: Purge contacts when an AoR is deleted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25339">ASTERISK-25339</a>: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc6fe07f5c114bdeaef4a3b83a11faaa9d1046eb">[bc6fe07f5c]</a> Matt Jordan -- res_pjsip/pjsip_configuration: Disregard empty auth values</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25304">ASTERISK-25304</a>: res_pjsip: XML sanitization may write past buffer<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8521a86367ac6090210a89878c8fee6d19c43642">[8521a86367]</a> Joshua Colp -- res_pjsip: Ensure sanitized XML is NULL terminated.</li>
</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25387">ASTERISK-25387</a>: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1dd0e220bf98ca93b825d7b5af4160f7718eab38">[1dd0e220bf]</a> Matt Jordan -- res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25306">ASTERISK-25306</a>: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c126afe18f9073f3ee74e45f574da421131b9fa2">[c126afe18f]</a> Richard Mudgett -- res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e25569ef95c6de6e9267df4673bd1d774b82a000">[e25569ef95]</a> Mark Michelson -- res_pjsip_pubsub: More accurately persist packet.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25356">ASTERISK-25356</a>: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b1561f4c854c37691bd24227b8f722d1dac4291">[1b1561f4c8]</a> Joshua Colp -- res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25297">ASTERISK-25297</a>: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13eb491e35ae6a99164dec6a62d7f05784c75c11">[13eb491e35]</a> Richard Mudgett -- res_pjsip_session.c: Fix crashes seen when call cancelled.</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25438">ASTERISK-25438</a>: res_rtp_asterisk: ICE role message even when ICE is not enabled<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9913d47697716040917b49c0c60ae2d98493d516">[9913d47697]</a> Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25318">ASTERISK-25318</a>: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2c73190825bf4c9cedb1031327199767a4a3ca8">[c2c7319082]</a> Joshua Colp -- res_pjsip_session: Don't invoke session supplements twice for BYE requests.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25292">ASTERISK-25292</a>: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=10ba72a9279c591e800ad2656d367b881f73203d">[10ba72a927]</a> Mark Michelson -- Add a test event for inband ringing.</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25310">ASTERISK-25310</a>: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ed9c9a280c08a17fce602c15d2b01de199ca736">[4ed9c9a280]</a> Guido Falsi -- Core/General: Add #ifdef needed on FreeBSD.</li>
</ul><br><h4>Category: Resources/res_ari_applications</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</li>
</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</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: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25299">ASTERISK-25299</a>: RTP port leaks with incoming OOH323 calls<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=480c443e2691272a7227e0949244e80e53bc31b2">[480c443e26]</a> Alexander Anikin -- chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25323">ASTERISK-25323</a>: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c94f46080f60435fffd197d14441ccf9d963521b">[c94f46080f]</a> Scott Griepentrog -- CHAOS: avoid crash if string create fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cc59533b903b3d55e8b388f28385287e712ae62">[4cc59533b9]</a> Richard Mudgett -- CHAOS: res_pjsip_diversion avoid crash if allocation fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb6b5c684b8772ba008339a417725a208f72409e">[fb6b5c684b]</a> Scott Griepentrog -- PJSIP: avoid crash when getting rtp peer</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f72f9ceefca52002c45f5910219dbcb0f9437a79">[f72f9ceefc]</a> Scott Griepentrog -- pjsip: avoid possible crash req_caps allocation failure</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6862c2a167f4ed2cb8511bb1ae94a13582afa25b">[6862c2a167]</a> Scott Griepentrog -- Chaos: handle failed allocation in get_media_encryption_type</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1cd6366588c66dce5be66541ceb7f828fde3773">[f1cd636658]</a> Scott Griepentrog -- Chaos: make hangup NULL tolerant</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab373f2ceffcad3a497663027199f4f4a81f644b">[ab373f2cef]</a> Scott Griepentrog -- CHAOS: prevent sorcery object with null id</li>
</ul><br><h4>Category: Resources/res_hep_rtcp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25352">ASTERISK-25352</a>: res_hep_rtcp correlation_id is different then res_hep<br/>Reported by: Kevin Scott Adams<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78d0b9d97ecf034468e252440217dd4bc371ef71">[78d0b9d97e]</a> Matt Jordan -- channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24602">ASTERISK-24602</a>: Unable to call WebRTC client via wss on chan_pjsip<br/>Reported by: Oleg Kozlov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d32e516c7cd48979db092a82b97a7ac4a743f526">[d32e516c7c]</a> Martin Tomec -- res/pjsip: Mark WSS transport as secure</li>
</ul><br><h3>Improvement</h3><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=cc0eff5651e792d9f97a433fbbac99992c20cb46">cc0eff5651</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8cd191b885c5410faa11e28b52d37fea9d5c5197">8cd191b885</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3777c24fd93fe757ce8c68daf9c2749938eea8f">a3777c24fd</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68121cef21b3753a28248b493d211a6d3176e67a">68121cef21</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d72dab4f402c06b304e339f4e8fd801b502cb003">d72dab4f40</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9da83dbd1524aa1c74907066ec6480e9d56651f9">9da83dbd15</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c60f9326ce609061316f4bc3b1522caa7282482">8c60f9326c</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=316d47755b0d6239cb5bbd674d7596046ad4cddd">316d47755b</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74a86d0a72c9858dc965e8e2b59ae4a1e0294033">74a86d0a72</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c39bea6f00560dd6bc52d9d793e1ae7ebf130b7">4c39bea6f0</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3521e9469f95b962e52edbbb6e0cdef6d3ceab4">c3521e9469</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a44f6aa046c988f3c1ce9defba5038f83255a4c4">a44f6aa046</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd74af7e460083df8ca774db2793791ff6413c6e">dd74af7e46</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a11a78ca34fae7ef2069d0a9ae6a5cf1fd2856d7">a11a78ca34</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=570329ec8a05c27675721734e2da3c42c707f0f7">570329ec8a</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51c9ff47f6bab1aed30f95a8601e0ddf253b657e">51c9ff47f6</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0fb436eda914dca26e96d304d3c9daca2be54de">a0fb436eda</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bba1c4066be4d752f106266a7c084ba2340c2777">bba1c4066b</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82c4aecdbb4466843a41e0591ed0e7c16c1902a6">82c4aecdbb</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc18db73884513e5636fc4af8d1b8bf2ba355c3a">bc18db7388</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9c53f95e38e7e9e47c9843bcbb0e8836cafed1b">b9c53f95e3</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d30939b6e84f18eb1fa3eb9819951fe8a1c764f4">d30939b6e8</a></td><td>Kevin Harwell</td><td>ARI: Changed version from 1.8.0 to 1.9.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f19c9baded56a5adb419a7bcb1ac00fbe09f404">5f19c9bade</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Fix user sort compare function.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a8576403968bd1f49c8bef67735c04d05fb6983">3a85764039</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Optimize conf_alloc() object init.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbeda190c3f05e95a82c7d9609c66dcf3ce35bd3">bbeda190c3</a></td><td>Richard Mudgett</td><td>app_dial.c: Remove some no-op code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe5077b1f8caed2df419e1bd7b872657b7def726">fe5077b1f8</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Eliminate race during initial NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c713fdf18ffa934e0cac8ddb29e4ad95a68200b">5c713fdf18</a></td><td>Mark Michelson</td><td>scheduler: Use queue for allocating sched IDs.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e75aff53e6ee68833595db101e43329adf9a4459">e75aff53e6</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d91d01df180a712485aa60d14fda2aa9e5063d2">4d91d01df1</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add some notification comments.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f36a9d122171781dd8a99d32792a0c19103b1f15">f36a9d1221</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94582f8fabb926d197ce0f3a01208b385975ec09">94582f8fab</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix off-nominal memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b3ed52239b24546b1ee12156dadccb70db7403e">8b3ed52239</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix one byte buffer overrun error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4329bd1e4c059e714122465901ea2c46dd924b71">4329bd1e4c</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a456a20ecf835bcf70ebc2a279e230df402bec08">a456a20ecf</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add missing error return in load_module().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f58f4c6e2762e5ad7eccf7065e63b345f4cda7f6">f58f4c6e27</a></td><td>Richard Mudgett</td><td>res_pjsip/location.c: Use the builtin ao2_callback() match function instead.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4eedd9ef9d7c000cd8d67cbeb1789ac6d71860aa">4eedd9ef9d</a></td><td>Matt Jordan</td><td>main/config_options: Check for existance of internal object before derefing</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=695f26cbb759f8ecc18c6b1d6cf84b3105b2f007">695f26cbb7</a></td><td>David M. Lee</td><td>res_rtp_asterisk: Add more ICE debugging</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61c6c6aa6c60636207567a49c6320946c1840e99">61c6c6aa6c</a></td><td>David M. Lee</td><td>Fix when remote candidates exceed PJ_ICE_MAX_CAND</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad9cb6c2ce6dbe9c985c6891daf53cc4160e3a13">ad9cb6c2ce</a></td><td>Mark Michelson</td><td>res_pjsip: Fix contact refleak on stateful responses.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c4d0c3506374b89502bd6c1bda89c3f241b6708">7c4d0c3506</a></td><td>Joshua Colp</td><td>res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0582776f7fe55e72acda586fb32185ad7879aeab">0582776f7f</a></td><td>Richard Mudgett</td><td>ari/ari_websockets.c: Fix ast_debug parameter type mismatch.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77518d54344945d7d3bfc1ebfe61d97704fa5dfa">77518d5434</a></td><td>Richard Mudgett</td><td>res_http_websocket.c: Fix some off nominal path cleanup.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c61547fee6fdc1e132d359311da48e87d98d25b1">c61547fee6</a></td><td>Richard Mudgett</td><td>res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd867cd0787b124984caf2604478212651ea4c03">bd867cd078</a></td><td>Richard Mudgett</td><td>app_queue.c: Extract some functions for simpler code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ded51e3d77a13eed059a85a083d0ab0324a77db7">ded51e3d77</a></td><td>Richard Mudgett</td><td>app_queue.c: Fix error checking in QUEUE_MEMBER() read.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b719f56c72c9cc66879eeef11de2ef4498cba648">b719f56c72</a></td><td>Mark Michelson</td><td>res_pjsip_sdp_rtp: Restore removed NULL check.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cea5dc7b8afd0e8cbde4c5d253bac3219125b168">cea5dc7b8a</a></td><td>Richard Mudgett</td><td>audiohook.c: Simplify variable usage in audiohook_read_frame_both().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e18c300550453df06507e32c4bc78ef91d369f27">e18c300550</a></td><td>Joshua Colp</td><td>res_http_websocket: When shutting down a session don't close closed socket</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e194047acb3ba846356041c1a6222caefc65a2e">8e194047ac</a></td><td>Matt Jordan</td><td>res/res_format_attr_silk: Expose format attributes to other modules</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0f451c35ed56b08353c4c3150bf847867f74fe7">a0f451c35e</a></td><td>Matt Jordan</td><td>main/format: Add an API call for retrieving format attributes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26f0559a94519b735396ed4ce90e23c1d9d5b332">26f0559a94</a></td><td>David M. Lee</td><td>Replace htobe64 with htonll</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27dc2094e98b59b8a50b059ddd6048285a42e6b9">27dc2094e9</a></td><td>Mark Michelson</td><td>res_http_websocket: Debug write lengths.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39cc28f6ea2140ad6d561fd4c9e9a66f065cecee">39cc28f6ea</a></td><td>Mark Michelson</td><td>res_http_websocket: Avoid passing strlen() to ast_websocket_write().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1519eb44a796911c7c438bbe4e31bb89be244387">1519eb44a7</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Must protect mime_types_len with mime_types_lock.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a93b7a927c5975c0a8889dc66868f81e4eef8aa3">a93b7a927c</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=741fa0d26d38f7a9ed98c595ee1bb6b6ce8a9923">741fa0d26d</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fixup some whitespace.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b21fd9a38bcd89402249440c1670ce48781f30">89b21fd9a3</a></td><td>Richard Mudgett</td><td>rtp_engine.h: No sense allowing payload types larger than RFC allows.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7427c7f13b1d0c9095e83a3ea38394f521d3a75e">7427c7f13b</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Minor tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e20f435b6093c2c72e678b9fad1ed037c3191b88">e20f435b60</a></td><td>Richard Mudgett</td><td>rtp_engine.h: Misc comment fixes.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc5d7f9c37fe8a4ff5744ab8620898ccae6a7d2a">bc5d7f9c37</a></td><td>Richard Mudgett</td><td>chan_sip.c: Tweak glue-&gt;update_peer() parameter nil value.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=48698a5e21d7307f61b5fb2bd39fd593bc1423ca">48698a5e21</a></td><td>Mark Michelson</td><td>res_http_websocket: Properly encode 64 bit payload</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-13.5.0-summary.html | 441 --
asterisk-13.5.0-summary.txt | 1141 ------
b/.version | 2
b/CHANGES | 35
b/ChangeLog | 1884 +++++++++-
b/Makefile | 1
b/UPGRADE.txt | 6
b/addons/chan_ooh323.c | 1
b/addons/ooh323c/src/ooq931.c | 6
b/apps/app_dial.c | 156
b/apps/app_page.c | 28
b/apps/app_queue.c | 412 +-
b/apps/app_record.c | 3
b/bridges/bridge_holding.c | 6
b/channels/chan_dahdi.c | 55
b/channels/chan_pjsip.c | 18
b/channels/chan_sip.c | 78
b/channels/chan_skinny.c | 26
b/channels/pjsip/dialplan_functions.c | 5
b/channels/sip/include/security_events.h | 3
b/channels/sip/security_events.c | 5
b/codecs/codec_gsm.c | 29
b/codecs/codec_ilbc.c | 28
b/codecs/codec_lpc10.c | 41
b/codecs/codec_speex.c | 60
b/configure | 63
b/configure.ac | 6
b/contrib/ast-db-manage/config/versions/154177371065_add_default_from_user.py | 22
b/contrib/realtime/mssql/mssql_config.sql | 8
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 8
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/astversion | 536 ++
b/contrib/scripts/install_prereq | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format.h | 23
b/include/asterisk/logger.h | 44
b/include/asterisk/res_pjsip.h | 40
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/rtp_engine.h | 18
b/include/asterisk/stasis_app.h | 15
b/main/audiohook.c | 43
b/main/bridge_channel.c | 7
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/dial.c | 25
b/main/endpoints.c | 3
b/main/format.c | 11
b/main/logger.c | 272 +
b/main/pbx.c | 146
b/main/rtp_engine.c | 153
b/main/sched.c | 143
b/main/sorcery.c | 4
b/main/stasis_endpoints.c | 78
b/main/taskprocessor.c | 12
b/main/translate.c | 55
b/main/utils.c | 4
b/res/ari/ari_model_validators.c | 445 ++
b/res/ari/ari_model_validators.h | 118
b/res/ari/ari_websockets.c | 9
b/res/ari/config.c | 72
b/res/ari/resource_asterisk.c | 127
b/res/ari/resource_asterisk.h | 63
b/res/ari/resource_events.c | 24
b/res/ari/resource_events.h | 2
b/res/parking/parking_applications.c | 65
b/res/res_ari.c | 19
b/res/res_ari_asterisk.c | 300 +
b/res/res_ari_events.c | 6
b/res/res_config_sqlite.c | 8
b/res/res_format_attr_silk.c | 24
b/res/res_http_websocket.c | 80
b/res/res_pjsip.c | 131
b/res/res_pjsip/config_global.c | 18
b/res/res_pjsip/location.c | 51
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/presence_xml.c | 31
b/res/res_pjsip_diversion.c | 4
b/res/res_pjsip_multihomed.c | 23
b/res/res_pjsip_nat.c | 2
b/res/res_pjsip_pidf_digium_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 80
b/res/res_pjsip_sdp_rtp.c | 70
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_t38.c | 15
b/res/res_pjsip_transport_websocket.c | 5
b/res/res_rtp_asterisk.c | 79
b/res/res_stasis.c | 45
b/res/res_stasis_device_state.c | 54
b/res/stasis/app.c | 335 +
b/res/stasis/app.h | 15
b/res/stasis/messaging.c | 44
b/rest-api-templates/ari_model_validators.c.mustache | 2
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 130
b/rest-api/api-docs/bridges.json | 2
b/rest-api/api-docs/channels.json | 2
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 109
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_core_format.c | 57
106 files changed, 6435 insertions(+), 2535 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.6.0
Date: 2015-10-09
<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-13.5.0.
----------------------------------------------------------------------
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
40 Richard Mudgett 1 Elazar Broad 13 Matt Jordan
18 Kevin Harwell 11 Joshua Colp
16 Joshua Colp 11 Richard Mudgett
15 Mark Michelson 10 Scott Griepentrog
13 Matt Jordan 8 Mark Michelson
9 Scott Griepentrog 8 Mark Michelson
8 Kevin Harwell 7 John Hardin
3 Scott Emidy 5 Kevin Harwell
3 David M. Lee 4 Scott Griepentrog
2 Alexander Traud 3 Richard Mudgett
2 Alexander Anikin 2 Kevin Harwell
2 Jonathan Rose 2 Alexander Traud
1 Martin Tomec 2 Alexandr Dranchuk
1 Elazar Broad 2 Dmitriy Serov
1 Walter Doekes 2 Stefan EngstrAP:m
1 Rodrigo RamArez Norambuena 1 Oleg Kozlov
1 Mark Duncan 1 Walter Doekes
1 Benjamin Ford 1 Etienne Lessard
1 Guido Falsi 1 Lorne Gaetz
1 Chet Stevens
1 Rodrigo Ramirez Norambuena
1 Ashley Sanders
1 Guido Falsi
1 Lorne Gaetz
1 Chet Stevens
1 Kevin Scott Adams
1 Elazar Broad
1 Sean Pimental
1 Jonathan Rose
1 Etienne Lessard
1 Alexandr Dranchuk
1 yaron nahum
1 Elazar Broad
1 yaron nahum
1 Guido Falsi
----------------------------------------------------------------------
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: Resources/res_ari
ASTERISK-25252: ARI: Add the ability to manipulate log channels
Reported by: Matt Jordan
* [df9ce36366] Scott Emidy -- ARI: Retrieve existing log channels
* [e9f1bc08cb] Scott Emidy -- ARI: Creating log channels
* [78364132ce] Scott Emidy -- ARI: Deleting log channels
* [1ae762634c] Benjamin Ford -- ARI: Rotate log channels.
Category: Resources/res_pjsip
ASTERISK-25377: res_pjsip: Change default "From user" from UUID to
something more palatable
Reported by: Mark Michelson
* [ac62928d6b] Mark Michelson -- res_pjsip: Change default from user
value.
Bug
Category: Addons/chan_ooh323
ASTERISK-25227: No audio at in-band announcements in ooh323 channel
Reported by: Alexandr Dranchuk
* [71408df2b8] Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE
with inband info available
Category: Applications/app_dial
ASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard Mudgett
* [6b1e7583c1] Richard Mudgett -- app_queue.c: Force COLP update if
outgoing channel name changed.
* [6bf304bf25] Richard Mudgett -- app_queue.c: Factor out a connected
line update routine.
* [e36b5f1e8e] Richard Mudgett -- app_dial.c: Make 'A' option pass COLP
updates.
* [747bfac895] Richard Mudgett -- app_dial.c: Force COLP update if
outgoing channel name changed.
* [14481d9aa0] Richard Mudgett -- app_dial.c: Factor out a connected
line update routine.
Category: Applications/app_mixmonitor
ASTERISK-25322: Crash occurs when using MixMonitor with t() or r()
options.
Reported by: Richard Mudgett
* [b3a56bee83] Richard Mudgett -- audiohook.c: Fix MixMonitor crash when
using the r() or t() options.
Category: Applications/app_page
ASTERISK-25384: Regular Asterisk crashes when using Page application.
"user_data is NULL"
Reported by: Chet Stevens
* [5f15cd93f0] Richard Mudgett -- app_page.c: Fix crash when forwarding
with a predial handler.
Category: Applications/app_queue
ASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard Mudgett
* [6b1e7583c1] Richard Mudgett -- app_queue.c: Force COLP update if
outgoing channel name changed.
* [6bf304bf25] Richard Mudgett -- app_queue.c: Factor out a connected
line update routine.
* [e36b5f1e8e] Richard Mudgett -- app_dial.c: Make 'A' option pass COLP
updates.
* [747bfac895] Richard Mudgett -- app_dial.c: Force COLP update if
outgoing channel name changed.
* [14481d9aa0] Richard Mudgett -- app_dial.c: Factor out a connected
line update routine.
ASTERISK-25399: app_queue: AgentComplete event has wrong reason
Reported by: Kevin Harwell
* [4fb95bbc4e] Kevin Harwell -- app_queue: AgentComplete event has wrong
reason
ASTERISK-25185: Segfault in app_queue on transfer scenarios
Reported by: Etienne Lessard
* [6409e7b11a] Kevin Harwell -- app_queue: Crash when transferring
ASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and
using PauseQueueMember
Reported by: Lorne Gaetz
* [e5f5b9f384] Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER
'paused' and 'ringinuse'.
Category: Applications/app_record
ASTERISK-25410: app_record: RECORDED_FILE variable not being populated
Reported by: Kevin Harwell
* [aeddee39fb] Kevin Harwell -- app_record: RECORDED_FILE variable not
being populated
Category: Bridges/bridge_holding
ASTERISK-25271: Parking & blind transfer: Transferer channel not hung up
if no MOH
Reported by: Kevin Harwell
* [8458b8d441] Jonathan Rose -- holding_bridge: ensure moh participants
get frames
Category: Channels/chan_dahdi
ASTERISK-25315: DAHDI channels send shortened duration DTMF tones.
Reported by: Richard Mudgett
* [256bc52b66] Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write
buffer after starting DTMF.
* [800e0ea48d] Richard Mudgett -- chan_dahdi.c: Lock private struct for
ast_write().
Category: Channels/chan_sip/CodecHandling
ASTERISK-25309: [patch] iLBC 20 advertised
Reported by: Alexander Traud
* [f68c995bc9] Alexander Traud -- chan_sip: Fix negotiation of iLBC 30.
Category: Channels/chan_sip/General
ASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on
call pickup
Reported by: Joshua Colp
* [c01111223f] Joshua Colp -- chan_sip: Allow call pickup to set the
hangup cause.
Category: Channels/chan_sip/Interoperability
ASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP
Reported by: Walter Doekes
* [b59c4d82b5] Walter Doekes -- chan_sip: Fix From header truncation for
extremely long CALLERID(name).
Category: Channels/chan_sip/Security Framework
ASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong
or non existent peer on invite
Reported by: Kevin Harwell
* [25af2d71c8] Kevin Harwell -- chan_sip.c: wrong peer searched in
sip_report_security_event
Category: Channels/chan_skinny
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Channels/chan_unistim
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Codecs/General
ASTERISK-25353: [patch] Transcoding while different in Frame size = Frames
lost
Reported by: Alexander Traud
* [b88c54fa4b] Alexander Traud -- translate: Fix transcoding while
different in frame size.
Category: Core/Bridging
ASTERISK-25341: bridge: Hangups may get lost when executing actions
Reported by: Joshua Colp
* [6c2dab1e88] Joshua Colp -- bridge: Kick channel from bridge if hung
up during action.
Category: Core/BuildSystem
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG
enabled
Reported by: yaron nahum
* [028033e5a8] Richard Mudgett -- res/ari/config.c: Fix conf_alloc()
object init.
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
- add ECDH support and fallback to prime256v1
Reported by: Stefan EngstrAP:m
* [9a12804e59] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
when enabling PFS.
* [aed068844c] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
Category: Core/General
ASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes
erroneous duplicate RTCP messages; other potential scheduling issues in
chan_sip/chan_skinny
Reported by: Matt Jordan
* [10e790f81a] Matt Jordan -- res/res_rtp_asterisk: Fix assignment after
ao2 decrement
* [754daeca0a] Matt Jordan -- Fix improper usage of scheduler exposed by
5c713fdf18f
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG
enabled
Reported by: yaron nahum
* [028033e5a8] Richard Mudgett -- res/ari/config.c: Fix conf_alloc()
object init.
ASTERISK-25418: On-hold channels redirected out of a bridge appear to
still be on hold
Reported by: Mark Michelson
* [629458d349] Mark Michelson -- Do not swallow frames on channels
leaving bridges.
ASTERISK-25355: sched: ast_sched_del may return prematurely due to
spurious wakeup
Reported by: Joshua Colp
* [85e1cb51b2] Joshua Colp -- sched: ast_sched_del may return
prematurely due to spurious wakeup
Category: Core/Logging
ASTERISK-25305: Dynamic logger channels can be added multiple times
Reported by: Mark Michelson
* [f050fa76eb] Mark Michelson -- logger: Prevent duplicate dynamic
channels from being added.
ASTERISK-25407: Asterisk fails to log to multiple syslog destinations
Reported by: Elazar Broad
* [ec514ad64d] Elazar Broad -- core/logging: Fix logging to more than
one syslog channel
Category: Core/PBX
ASTERISK-25394: pbx: Incorrect device and presence state when changing
hint details
Reported by: Joshua Colp
* [2bd27d1222] Joshua Colp -- pbx: Update device and presence state when
changing a hint extension.
ASTERISK-25367: pbx: Long pattern match hints may cause "core show hints"
to crash
Reported by: Joshua Colp
* [cc1363209e] Joshua Colp -- pbx: Fix crash when issuing "core show
hints" with long pattern match.
ASTERISK-25362: Deadlock due to presence state callback
Reported by: Mark Michelson
* [03fe79f29e] Mark Michelson -- Fix deadlock on presence state changes.
Category: Core/RTP
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Resources/res_ari
ASTERISK-25325: ARI PUT reload chan_sip HTTP response 404
Reported by: Rodrigo Ramirez Norambuena
* [865377fc38] Rodrigo RamArez Norambuena -- chan_sip.c: Validation on
module reload
Category: Resources/res_http_websocket
ASTERISK-25312: res_http_websocket: Terminate connection on fatal cases
Reported by: Joshua Colp
* [b4e9416138] Joshua Colp -- res_http_websocket: Forcefully terminate
on write errors.
Category: Resources/res_parking
ASTERISK-25369: res_parking: ParkAndAnnounce - Inheritable variables
aren't applied to the announcer channel
Reported by: Jonathan Rose
* [fbf720db91] Jonathan Rose -- ParkAndAnnounce: Add variable
inheritance
Category: Resources/res_pjsip
ASTERISK-25295: res_pjsip crash - pjsip_uri_get_uri at
/usr/include/pjsip/sip_uri.h
Reported by: Dmitriy Serov
* [5469caa9dd] Joshua Colp -- res_pjsip: Use hash for contact object
identity instead of Contact URI.
* [a676ba2aad] Joshua Colp -- taskprocessor: Fix race condition between
unreferencing and finding.
ASTERISK-25381: res_pjsip: AoRs deleted via ARI (or other mechanism) do
not destroy their related contacts
Reported by: Matt Jordan
* [c3e6debdb9] Matt Jordan -- res/res_pjsip: Purge contacts when an AoR
is deleted
ASTERISK-25339: res_pjsip: Empty "auth" sections from non-config
backgrounds are interpreted as valid
Reported by: Matt Jordan
* [bc6fe07f5c] Matt Jordan -- res_pjsip/pjsip_configuration: Disregard
empty auth values
ASTERISK-25304: res_pjsip: XML sanitization may write past buffer
Reported by: Joshua Colp
* [8521a86367] Joshua Colp -- res_pjsip: Ensure sanitized XML is NULL
terminated.
Category: Resources/res_pjsip_nat
ASTERISK-25387: res_pjsip_nat: Malformed REGISTER request causes NAT'd
Contact header to not be rewritten
Reported by: Matt Jordan
* [1dd0e220bf] Matt Jordan -- res/res_pjsip_nat: Ignore REGISTER
requests when looking for a Record-Route
Category: Resources/res_pjsip_pubsub
ASTERISK-25306: Persistent subscriptions can save multiple SIP messages at
once, leading to potential crashes.
Reported by: Mark Michelson
* [c126afe18f] Richard Mudgett -- res_pjsip.c: Fix crash from corrupt
saved SUBSCRIBE message.
* [e25569ef95] Mark Michelson -- res_pjsip_pubsub: More accurately
persist packet.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-25356: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may
exist
Reported by: Joshua Colp
* [1b1561f4c8] Joshua Colp -- res_pjsip_sdp_rtp: Fix multiple keepalive
scheduled items.
Category: Resources/res_pjsip_session
ASTERISK-25297: Crashes running
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
Reported by: Richard Mudgett
* [13eb491e35] Richard Mudgett -- res_pjsip_session.c: Fix crashes seen
when call cancelled.
Category: Resources/res_rtp_asterisk
ASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not
enabled
Reported by: Joshua Colp
* [9913d47697] Joshua Colp -- res_rtp_asterisk: Move "Set role" warning
to be debug.
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
- add ECDH support and fallback to prime256v1
Reported by: Stefan EngstrAP:m
* [9a12804e59] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
when enabling PFS.
* [aed068844c] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
Category: Tests/testsuite
ASTERISK-25318:
tests/rest_api/applications/subscribe-endpoint/nominal/resource:
Sporadically failing
Reported by: Joshua Colp
* [c2c7319082] Joshua Colp -- res_pjsip_session: Don't invoke session
supplements twice for BYE requests.
ASTERISK-25292: Testuite:
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
Reported by: Kevin Harwell
* [10ba72a927] Mark Michelson -- Add a test event for inband ringing.
Improvement
Category: Core/General
ASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to
PTHREAD_EXPLICIT_SCHED
Reported by: Guido Falsi
* [4ed9c9a280] Guido Falsi -- Core/General: Add #ifdef needed on
FreeBSD.
Category: Resources/res_ari_applications
ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan
* [90165e306d] Matt Jordan -- res/res_stasis: Fix accidental
subscription to 'all' bridge topic
* [b50e372394] Matt Jordan -- ARI: Add events for Contact and Peer
Status changes
* [3502c0431d] Matt Jordan -- res/res_stasis_device_state: Allow for
subscribing to 'all' device state
* [4c9f613309] Matt Jordan -- ARI: Add the ability to subscribe to all
events
Category: Resources/res_ari_bridges
ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan
* [90165e306d] Matt Jordan -- res/res_stasis: Fix accidental
subscription to 'all' bridge topic
* [b50e372394] Matt Jordan -- ARI: Add events for Contact and Peer
Status changes
* [3502c0431d] Matt Jordan -- res/res_stasis_device_state: Allow for
subscribing to 'all' device state
* [4c9f613309] Matt Jordan -- ARI: Add the ability to subscribe to all
events
----------------------------------------------------------------------
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: Addons/chan_ooh323
ASTERISK-25299: RTP port leaks with incoming OOH323 calls
Reported by: Alexandr Dranchuk
* [480c443e26] Alexander Anikin -- chan_ooh323: call
ast_rtp_instance_stop on ooh323_destroy
Category: General
ASTERISK-25323: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG
Reported by: Scott Griepentrog
* [c94f46080f] Scott Griepentrog -- CHAOS: avoid crash if string create
fails
* [4cc59533b9] Richard Mudgett -- CHAOS: res_pjsip_diversion avoid crash
if allocation fails
* [fb6b5c684b] Scott Griepentrog -- PJSIP: avoid crash when getting rtp
peer
* [f72f9ceefc] Scott Griepentrog -- pjsip: avoid possible crash req_caps
allocation failure
* [6862c2a167] Scott Griepentrog -- Chaos: handle failed allocation in
get_media_encryption_type
* [f1cd636658] Scott Griepentrog -- Chaos: make hangup NULL tolerant
* [ab373f2cef] Scott Griepentrog -- CHAOS: prevent sorcery object with
null id
Category: Resources/res_hep_rtcp
ASTERISK-25352: res_hep_rtcp correlation_id is different then res_hep
Reported by: Kevin Scott Adams
* [78d0b9d97e] Matt Jordan -- channels/pjsip/dialplan_functions: Add an
option for extracting the SIP call-id
Category: pjproject/pjsip
ASTERISK-24602: Unable to call WebRTC client via wss on chan_pjsip
Reported by: Oleg Kozlov
* [d32e516c7c] Martin Tomec -- res/pjsip: Mark WSS transport as secure
Improvement
----------------------------------------------------------------------
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 |
|------------+-----------------+-----------------------------------------|
| cc0eff5651 | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| 8cd191b885 | Kevin Harwell | .version: Update for 13.6.0 |
|------------+-----------------+-----------------------------------------|
| a3777c24fd | Kevin Harwell | .lastclean: Update for 13.6.0 |
|------------+-----------------+-----------------------------------------|
| 68121cef21 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0 |
|------------+-----------------+-----------------------------------------|
| d72dab4f40 | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 9da83dbd15 | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 8c60f9326c | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| 316d47755b | Kevin Harwell | .version: Update for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 74a86d0a72 | Kevin Harwell | .lastclean: Update for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 4c39bea6f0 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| c3521e9469 | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| a44f6aa046 | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| dd74af7e46 | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| a11a78ca34 | Kevin Harwell | .version: Update for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| 570329ec8a | Kevin Harwell | .lastclean: Update for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| 51c9ff47f6 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| a0fb436eda | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| bba1c4066b | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| 82c4aecdbb | Kevin Harwell | .version: Update for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| bc18db7388 | Kevin Harwell | .lastclean: Update for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| b9c53f95e3 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| d30939b6e8 | Kevin Harwell | ARI: Changed version from 1.8.0 to |
| | | 1.9.0 |
|------------+-----------------+-----------------------------------------|
| 5f19c9bade | Richard Mudgett | res/ari/config.c: Fix user sort compare |
| | | function. |
|------------+-----------------+-----------------------------------------|
| 3a85764039 | Richard Mudgett | res/ari/config.c: Optimize conf_alloc() |
| | | object init. |
|------------+-----------------+-----------------------------------------|
| bbeda190c3 | Richard Mudgett | app_dial.c: Remove some no-op code. |
|------------+-----------------+-----------------------------------------|
| fe5077b1f8 | Mark Michelson | res_pjsip_pubsub: Eliminate race during |
| | | initial NOTIFY. |
|------------+-----------------+-----------------------------------------|
| 5c713fdf18 | Mark Michelson | scheduler: Use queue for allocating |
| | | sched IDs. |
|------------+-----------------+-----------------------------------------|
| | | res_pjsip_pubsub.c: Mark |
| e75aff53e6 | Richard Mudgett | ast_sip_create_subscription() as not |
| | | used. |
|------------+-----------------+-----------------------------------------|
| 4d91d01df1 | Richard Mudgett | res_pjsip_pubsub.c: Add some |
| | | notification comments. |
|------------+-----------------+-----------------------------------------|
| f36a9d1221 | Richard Mudgett | res_pjsip_pubsub.c: Set dlg_status code |
| | | instead of sending SIP response. |
|------------+-----------------+-----------------------------------------|
| 94582f8fab | Richard Mudgett | res_pjsip_pubsub.c: Fix off-nominal |
| | | memory leak. |
|------------+-----------------+-----------------------------------------|
| 8b3ed52239 | Richard Mudgett | res_pjsip_pubsub.c: Fix one byte buffer |
| | | overrun error. |
|------------+-----------------+-----------------------------------------|
| 4329bd1e4c | Richard Mudgett | res_pjsip_pubsub.c: Use ast_alloca() |
| | | instead of alloca(). |
|------------+-----------------+-----------------------------------------|
| a456a20ecf | Richard Mudgett | res_pjsip_pubsub.c: Add missing error |
| | | return in load_module(). |
|------------+-----------------+-----------------------------------------|
| f58f4c6e27 | Richard Mudgett | res_pjsip/location.c: Use the builtin |
| | | ao2_callback() match function instead. |
|------------+-----------------+-----------------------------------------|
| | | main/config_options: Check for |
| 4eedd9ef9d | Matt Jordan | existance of internal object before |
| | | derefing |
|------------+-----------------+-----------------------------------------|
| 695f26cbb7 | David M. Lee | res_rtp_asterisk: Add more ICE |
| | | debugging |
|------------+-----------------+-----------------------------------------|
| 61c6c6aa6c | David M. Lee | Fix when remote candidates exceed |
| | | PJ_ICE_MAX_CAND |
|------------+-----------------+-----------------------------------------|
| ad9cb6c2ce | Mark Michelson | res_pjsip: Fix contact refleak on |
| | | stateful responses. |
|------------+-----------------+-----------------------------------------|
| 7c4d0c3506 | Joshua Colp | res_pjsip_pubsub: On recreated notify |
| | | fail deleted sub_tree is referenced |
|------------+-----------------+-----------------------------------------|
| 0582776f7f | Richard Mudgett | ari/ari_websockets.c: Fix ast_debug |
| | | parameter type mismatch. |
|------------+-----------------+-----------------------------------------|
| 77518d5434 | Richard Mudgett | res_http_websocket.c: Fix some off |
| | | nominal path cleanup. |
|------------+-----------------+-----------------------------------------|
| c61547fee6 | Richard Mudgett | res_ari.c: Add missing off nominal |
| | | unlock and remove a RAII_VAR(). |
|------------+-----------------+-----------------------------------------|
| bd867cd078 | Richard Mudgett | app_queue.c: Extract some functions for |
| | | simpler code. |
|------------+-----------------+-----------------------------------------|
| ded51e3d77 | Richard Mudgett | app_queue.c: Fix error checking in |
| | | QUEUE_MEMBER() read. |
|------------+-----------------+-----------------------------------------|
| b719f56c72 | Mark Michelson | res_pjsip_sdp_rtp: Restore removed NULL |
| | | check. |
|------------+-----------------+-----------------------------------------|
| cea5dc7b8a | Richard Mudgett | audiohook.c: Simplify variable usage in |
| | | audiohook_read_frame_both(). |
|------------+-----------------+-----------------------------------------|
| e18c300550 | Joshua Colp | res_http_websocket: When shutting down |
| | | a session don't close closed socket |
|------------+-----------------+-----------------------------------------|
| 8e194047ac | Matt Jordan | res/res_format_attr_silk: Expose format |
| | | attributes to other modules |
|------------+-----------------+-----------------------------------------|
| a0f451c35e | Matt Jordan | main/format: Add an API call for |
| | | retrieving format attributes |
|------------+-----------------+-----------------------------------------|
| 26f0559a94 | David M. Lee | Replace htobe64 with htonll |
|------------+-----------------+-----------------------------------------|
| 27dc2094e9 | Mark Michelson | res_http_websocket: Debug write |
| | | lengths. |
|------------+-----------------+-----------------------------------------|
| 39cc28f6ea | Mark Michelson | res_http_websocket: Avoid passing |
| | | strlen() to ast_websocket_write(). |
|------------+-----------------+-----------------------------------------|
| 1519eb44a7 | Richard Mudgett | rtp_engine.c: Must protect |
| | | mime_types_len with mime_types_lock. |
|------------+-----------------+-----------------------------------------|
| a93b7a927c | Richard Mudgett | res_pjsip_sdp_rtp.c: Fix processing |
| | | wrong SDP media list. |
|------------+-----------------+-----------------------------------------|
| 741fa0d26d | Richard Mudgett | res_pjsip_sdp_rtp.c: Fixup some |
| | | whitespace. |
|------------+-----------------+-----------------------------------------|
| 89b21fd9a3 | Richard Mudgett | rtp_engine.h: No sense allowing payload |
| | | types larger than RFC allows. |
|------------+-----------------+-----------------------------------------|
| 7427c7f13b | Richard Mudgett | rtp_engine.c: Minor tweaks. |
|------------+-----------------+-----------------------------------------|
| e20f435b60 | Richard Mudgett | rtp_engine.h: Misc comment fixes. |
|------------+-----------------+-----------------------------------------|
| bc5d7f9c37 | Richard Mudgett | chan_sip.c: Tweak glue->update_peer() |
| | | parameter nil value. |
|------------+-----------------+-----------------------------------------|
| 48698a5e21 | Mark Michelson | res_http_websocket: Properly encode 64 |
| | | bit payload |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-13.5.0-summary.html | 441 --
asterisk-13.5.0-summary.txt | 1141 ------
b/.version | 2
b/CHANGES | 35
b/ChangeLog | 1884 +++++++++-
b/Makefile | 1
b/UPGRADE.txt | 6
b/addons/chan_ooh323.c | 1
b/addons/ooh323c/src/ooq931.c | 6
b/apps/app_dial.c | 156
b/apps/app_page.c | 28
b/apps/app_queue.c | 412 +-
b/apps/app_record.c | 3
b/bridges/bridge_holding.c | 6
b/channels/chan_dahdi.c | 55
b/channels/chan_pjsip.c | 18
b/channels/chan_sip.c | 78
b/channels/chan_skinny.c | 26
b/channels/pjsip/dialplan_functions.c | 5
b/channels/sip/include/security_events.h | 3
b/channels/sip/security_events.c | 5
b/codecs/codec_gsm.c | 29
b/codecs/codec_ilbc.c | 28
b/codecs/codec_lpc10.c | 41
b/codecs/codec_speex.c | 60
b/configure | 63
b/configure.ac | 6
b/contrib/ast-db-manage/config/versions/154177371065_add_default_from_user.py | 22
b/contrib/realtime/mssql/mssql_config.sql | 8
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 8
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/astversion | 536 ++
b/contrib/scripts/install_prereq | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format.h | 23
b/include/asterisk/logger.h | 44
b/include/asterisk/res_pjsip.h | 40
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/rtp_engine.h | 18
b/include/asterisk/stasis_app.h | 15
b/main/audiohook.c | 43
b/main/bridge_channel.c | 7
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/dial.c | 25
b/main/endpoints.c | 3
b/main/format.c | 11
b/main/logger.c | 272 +
b/main/pbx.c | 146
b/main/rtp_engine.c | 153
b/main/sched.c | 143
b/main/sorcery.c | 4
b/main/stasis_endpoints.c | 78
b/main/taskprocessor.c | 12
b/main/translate.c | 55
b/main/utils.c | 4
b/res/ari/ari_model_validators.c | 445 ++
b/res/ari/ari_model_validators.h | 118
b/res/ari/ari_websockets.c | 9
b/res/ari/config.c | 72
b/res/ari/resource_asterisk.c | 127
b/res/ari/resource_asterisk.h | 63
b/res/ari/resource_events.c | 24
b/res/ari/resource_events.h | 2
b/res/parking/parking_applications.c | 65
b/res/res_ari.c | 19
b/res/res_ari_asterisk.c | 300 +
b/res/res_ari_events.c | 6
b/res/res_config_sqlite.c | 8
b/res/res_format_attr_silk.c | 24
b/res/res_http_websocket.c | 80
b/res/res_pjsip.c | 131
b/res/res_pjsip/config_global.c | 18
b/res/res_pjsip/location.c | 51
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/presence_xml.c | 31
b/res/res_pjsip_diversion.c | 4
b/res/res_pjsip_multihomed.c | 23
b/res/res_pjsip_nat.c | 2
b/res/res_pjsip_pidf_digium_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 80
b/res/res_pjsip_sdp_rtp.c | 70
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_t38.c | 15
b/res/res_pjsip_transport_websocket.c | 5
b/res/res_rtp_asterisk.c | 79
b/res/res_stasis.c | 45
b/res/res_stasis_device_state.c | 54
b/res/stasis/app.c | 335 +
b/res/stasis/app.h | 15
b/res/stasis/messaging.c | 44
b/rest-api-templates/ari_model_validators.c.mustache | 2
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 130
b/rest-api/api-docs/bridges.json | 2
b/rest-api/api-docs/channels.json | 2
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 109
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_core_format.c | 57
106 files changed, 6435 insertions(+), 2535 deletions(-)

View File

@@ -3199,8 +3199,8 @@ static int ast_rtcp_write(const void *data)
/*
* Not being rescheduled.
*/
ao2_ref(instance, -1);
rtp->rtcp->schedid = -1;
ao2_ref(instance, -1);
}
return res;