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19 Commits

Author SHA1 Message Date
mmichelson
7032890c9e ChangeLog: Updated for 13.7.2 2016-02-05 15:32:40 -05:00
mmichelson
736d0671b6 Release summaries: Add summaries for 13.7.2 2016-02-05 15:32:36 -05:00
Mark Michelson
0ba44bd6f3 Release summaries: Remove previous versions 2016-02-05 14:32:13 -06:00
mmichelson
b750dfe202 .version: Update for 13.7.2 2016-02-05 15:32:13 -05:00
mmichelson
c1b94ffe78 .lastclean: Update for 13.7.2 2016-02-05 15:32:13 -05:00
mmichelson
932ed1ab5b realtime: Add database scripts for 13.7.2 2016-02-05 15:32:13 -05:00
Mark Michelson
8de94229ba res_sorcery_realtime: Fix regex regression.
A regression was introduced where searching for realtime PJSIP objects
by regex by starting the regex with a leading "^" would cause no items
to be returned.

This was due to a change which attempted to drop the requirement for a
leading "^" to be present due to how some CLI commands formulate their
regexes. However, the change, rather than simply eliminating the
requirement, caused any regexes that did begin with "^" to end up not
returning the expected results.

This change fixes the problem by inspecting the regex and formulating
the realtime query differently depending on if it begins with "^".

ASTERISK-25702 #close
Reported by Nic Colledge

Patches:
    realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691

Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693
(cherry picked from commit 32fc784284)
2016-02-05 09:40:49 -06:00
Mark Michelson
a56f55d566 Check for OpenSSL defines before trying to use them.
The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.

This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.

Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
2016-02-05 09:39:03 -06:00
kharwell
211d2836de ChangeLog: Updated for 13.7.1 2016-02-03 16:33:58 -05:00
kharwell
20a93c6357 Release summaries: Add summaries for 13.7.1 2016-02-03 16:33:57 -05:00
Kevin Harwell
d99d10bd4c Release summaries: Remove previous versions 2016-02-03 15:33:43 -06:00
kharwell
e937a6db11 .version: Update for 13.7.1 2016-02-03 16:33:43 -05:00
kharwell
30a6826f1e .lastclean: Update for 13.7.1 2016-02-03 16:33:43 -05:00
kharwell
1c79fa7eb8 realtime: Add database scripts for 13.7.1 2016-02-03 16:33:43 -05:00
Kevin Harwell
b796a07ceb Merge "AST-2016-003 udptl.c: Fix uninitialized values." into 13.7 2016-02-03 15:18:34 -06:00
Kevin Harwell
9aacccf04a Merge "AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow." into 13.7 2016-02-03 15:15:52 -06:00
Joshua Colp
1e7854dfa2 AST-2016-001 http: Provide greater control of TLS and set modern defaults.
This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.

The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.

ASTERISK-24972 #close

Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
2016-02-03 15:09:15 -06:00
Richard Mudgett
b0646ff0da AST-2016-003 udptl.c: Fix uninitialized values.
Sending UDPTL packets to Asterisk with the right amount of missing
sequence numbers and enough redundant 0-length IFP packets, can make
Asterisk crash.

ASTERISK-25603 #close
Reported by: Walter Doekes

ASTERISK-25742 #close
Reported by: Torrey Searle

Change-Id: I97df8375041be986f3f266ac1946a538023a5255
2016-02-03 15:07:25 -06:00
Richard Mudgett
f08083f0f0 AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow.
Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times.  These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.

NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.

* The overflow is now detected and the previous timeout time is
calculated.

ASTERISK-25397 #close
Reported by: Alexander Traud

Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
2016-02-03 15:05:24 -06:00
16 changed files with 484 additions and 1522 deletions

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@@ -1 +1 @@
13.7.0
13.7.2

131
ChangeLog
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@@ -1,3 +1,134 @@
2016-02-05 20:32 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.7.2 Released.
2016-02-05 14:32 +0000 [0ba44bd6f3] Mark Michelson <mmichelson@lunkwill.digium.internal>
* Release summaries: Remove previous versions
2016-02-05 14:32 +0000 [b750dfe202] Mark Michelson <mmichelson@lunkwill>
* .version: Update for 13.7.2
2016-02-05 14:32 +0000 [c1b94ffe78] Mark Michelson <mmichelson@lunkwill>
* .lastclean: Update for 13.7.2
2016-02-05 14:32 +0000 [932ed1ab5b] Mark Michelson <mmichelson@lunkwill>
* realtime: Add database scripts for 13.7.2
2016-02-02 10:52 +0000 [8de94229ba] Alexei Gradinari License #5691
* res_sorcery_realtime: Fix regex regression.
A regression was introduced where searching for realtime PJSIP objects
by regex by starting the regex with a leading "^" would cause no items
to be returned.
This was due to a change which attempted to drop the requirement for a
leading "^" to be present due to how some CLI commands formulate their
regexes. However, the change, rather than simply eliminating the
requirement, caused any regexes that did begin with "^" to end up not
returning the expected results.
This change fixes the problem by inspecting the regex and formulating
the realtime query differently depending on if it begins with "^".
ASTERISK-25702 #close
Reported by Nic Colledge
Patches:
realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691
Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693
(cherry picked from commit 32fc784284b570a05841d95c6d9a373b4bf3a35d)
2016-02-04 16:17 +0000 [a56f55d566] Mark Michelson <mmichelson@digium.com>
* Check for OpenSSL defines before trying to use them.
The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.
This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.
Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
2016-02-03 21:33 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.7.1 Released.
2016-02-03 15:33 +0000 [d99d10bd4c] Kevin Harwell <kharwell@lunkwill.digium.internal>
* Release summaries: Remove previous versions
2016-02-03 15:33 +0000 [e937a6db11] Kevin Harwell <kharwell@lunkwill>
* .version: Update for 13.7.1
2016-02-03 15:33 +0000 [30a6826f1e] Kevin Harwell <kharwell@lunkwill>
* .lastclean: Update for 13.7.1
2016-02-03 15:33 +0000 [1c79fa7eb8] Kevin Harwell <kharwell@lunkwill>
* realtime: Add database scripts for 13.7.1
2016-02-03 12:05 +0000 [1e7854dfa2] Joshua Colp <jcolp@digium.com>
* AST-2016-001 http: Provide greater control of TLS and set modern defaults.
This change exposes the configuration of various aspects of the TLS
support and sets the default to the modern standards.
The TLS cipher is now set to the best values according to the
Mozilla OpSec team, different TLS versions can now be disabled, and
the cipher order can be forced to be that of the server instead of
the client.
ASTERISK-24972 #close
Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8
2015-12-07 12:46 +0000 [b0646ff0da] Richard Mudgett <rmudgett@digium.com>
* AST-2016-003 udptl.c: Fix uninitialized values.
Sending UDPTL packets to Asterisk with the right amount of missing
sequence numbers and enough redundant 0-length IFP packets, can make
Asterisk crash.
ASTERISK-25603 #close
Reported by: Walter Doekes
ASTERISK-25742 #close
Reported by: Torrey Searle
Change-Id: I97df8375041be986f3f266ac1946a538023a5255
2015-09-28 17:07 +0000 [f08083f0f0] Richard Mudgett <rmudgett@digium.com>
* AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow.
Setting the sip.conf timert1 value to a value higher than 1245 can cause
an integer overflow and result in large retransmit timeout times. These
large timeout times hold system file descriptors hostage and can cause the
system to run out of file descriptors.
NOTE: The default sip.conf timert1 value is 500 which does not expose the
vulnerability.
* The overflow is now detected and the previous timeout time is
calculated.
ASTERISK-25397 #close
Reported by: Alexander Traud
Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290
2016-01-15 19:01 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 13.7.0 Released.

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@@ -1,423 +0,0 @@
<!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-13.7.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.7.0</h3><h3 align="center">Date: 2016-01-15</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-13.6.0.</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%">33 Kevin Harwell <kharwell@lunkwill><br/>23 Matt Jordan <mjordan@digium.com><br/>23 Richard Mudgett <rmudgett@digium.com><br/>15 Mark Michelson <mmichelson@digium.com><br/>14 Alexander Traud <pabstraud@compuserve.com><br/>11 Joshua Colp <jcolp@digium.com><br/>8 gtjoseph <george.joseph@fairview5.com><br/>6 Kevin Harwell <kharwell@lunkwill.digium.internal><br/>5 Corey Farrell <git@cfware.com><br/>4 Walter Doekes <walter+asterisk@wjd.nu><br/>4 Jonathan Rose <jrose@digium.com><br/>3 Ivan Poddubny <ivan.poddubny@gmail.com><br/>2 Tyler Cambron <tcambron@digium.com><br/>1 Eugene Voityuk <eugene@thirdlane.com><br/>1 Filip Jenicek <phill@janevim.cz><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Alec Davis <sivad.a@paradise.net.nz><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Florian Sauerteig <ffs@ccn.net><br/>1 Niklas Larsson <niklas@tese.se><br/>1 server-pandora <server-pandora@xencall.com><br/>1 Olle Johansson (License 5267)<br/>1 Debian Amtelco <dan@amtelco.com><br/>1 Sean Bright (license #5060)<br/>1 David M. Lee <dlee@respoke.io><br/>1 Steve Davies <steve@one47.co.uk><br/></td><td width="33%">8 gtjoseph<br/>1 Dan Cropp<br/>1 starting asterisk -c until the colors stopped<br/>1 Alexander Traud<br/></td><td width="33%">13 Alexander Traud <pabstraud@compuserve.com><br/>12 Matt Jordan <mjordan@digium.com><br/>8 Joshua Colp <jcolp@digium.com><br/>7 George Joseph <george.joseph@fairview5.com><br/>7 gtjoseph<br/>4 Corey Farrell <git@cfware.com><br/>4 Michael Keuter <lists@mksolutions.info><br/>3 Walter Doekes <walter+asterisk@wjd.nu><br/>3 Richard Mudgett <rmudgett@digium.com><br/>2 Andrew Nagy<br/>2 Kevin Harwell <kharwell@digium.com><br/>2 Jonathan Rose <jrose@digium.com><br/>2 Andrew Nagy <andrew.nagy@the159.com><br/>2 Mark Michelson<br/>2 Mark Michelson <mmichelson@digium.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Steve Davies <steve@one47.co.uk><br/>1 ffs <ffs@ccn.net><br/>1 John Bigelow <jbigelow@digium.com><br/>1 Dmitriy Serov<br/>1 Chet Stevens <cwstevens@interact.ccsd.net><br/>1 Sean Pimental<br/>1 Niklas Larsson <niklas@tese.se><br/>1 Tyler Cambron <tcambron@digium.com><br/>1 Ben Langfeld<br/>1 Ashley Sanders<br/>1 Krzysztof Trempala<br/>1 Ashley Sanders <asanders@digium.com><br/>1 PowerPBX <canuck15@hotmail.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Dudás József <jozsef.dudas@gmail.com><br/>1 Hiroaki Komatsu <komatsu.hiroaki@po.ntts.co.jp><br/>1 Bryant Zimmerman <bryantz@zktech.com><br/>1 Jonathan Rose<br/>1 Ben Langfeld <ben@langfeld.me><br/>1 Rusty Newton<br/>1 Marcelo Terres<br/>1 Bojan Nemčić<br/>1 Chet Stevens<br/>1 Krzysztof Trempala <k.trempala@slican.pl><br/>1 Badalian Vyacheslav <slavon.net@gmail.com><br/>1 Olle Johansson <oej@edvina.net><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Taylor Hawkes <th71852@gmail.com><br/>1 Aleksei Kulakov <each.nir.vine@gmail.com><br/>1 Marcelo Terres <mhterres@gmail.com><br/>1 Badalian Vyacheslav<br/>1 Olle Johansson<br/>1 Filip Jenicek <phill@janevim.cz><br/>1 Dade Brandon <dade@xencall.com><br/>1 Dmitriy Serov <serov.d.p@gmail.com><br/>1 Bojan Nemčić <bojan.nemcic@voxdiversa.hr><br/>1 Alec Davis <sivad.a@paradise.net.nz><br/>1 John Bigelow<br/>1 Bryant Zimmerman<br/>1 dea <dan_austin@fitawi.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>New Feature</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25419">ASTERISK-25419</a>: Dialplan Application for Integration of StatsD<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e0040b88f83688e67d71521177cf4fa962bf32a">[1e0040b88f]</a> Tyler Cambron -- StatsD: Add res_statsd compatibility</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24922">ASTERISK-24922</a>: ARI: Add the ability to intercept hold and raise an event<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72cbb6df55bad972bf12800946f3c0b219aca049">[72cbb6df55]</a> Matt Jordan -- funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25419">ASTERISK-25419</a>: Dialplan Application for Integration of StatsD<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e0040b88f83688e67d71521177cf4fa962bf32a">[1e0040b88f]</a> Tyler Cambron -- StatsD: Add res_statsd compatibility</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25640">ASTERISK-25640</a>: pbx: Deadlock on features reload and state change hint.<br/>Reported by: Krzysztof Trempala<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20b22635dcc402e6b1cc39f796ea9e952e0da7ef">[20b22635dc]</a> Kevin Harwell -- pbx: Deadlock between contexts container and context_merge locks</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24958">ASTERISK-24958</a>: Forwarding loop detection inhibits certain desirable scenarios<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7debb986a51937886392f7f444f29483528f94ec">[7debb986a5]</a> Alec Davis -- app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25569">ASTERISK-25569</a>: app_meetme: Audio quality issues<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddf4dddf4ffa4866806f5ae6d5bc6cfaa9aa6fdb">[ddf4dddf4f]</a> Corey Farrell -- app_meetme: Set default value for audio_buffers.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25561">ASTERISK-25561</a>: app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!<br/>Reported by: Alec Davis<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7debb986a51937886392f7f444f29483528f94ec">[7debb986a5]</a> Alec Davis -- app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24779">ASTERISK-24779</a>: Passthrough OPUS codec not working with chan_pjsip<br/>Reported by: PowerPBX<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=33752e0837122fa73551dd3c424765477455b433">[33752e0837]</a> Sean Bright -- res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25404">ASTERISK-25404</a>: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=426263a64dc5dc80a51b6950ee0cb6b46f5f052c">[426263a64d]</a> Richard Mudgett -- chan_pjsip: Fix crash on reINVITE before initial INVITE completes.</li>
</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25160">ASTERISK-25160</a>: [patch] Opus Codec: SIP/SDP line fmtp missing when called internally<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d8d3991390bcae96dd5e33c59dcfe17e9932b7d4">[d8d3991390]</a> Alexander Traud -- format: Register format-attribute module with cached formats.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24543">ASTERISK-24543</a>: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs<br/>Reported by: Taylor Hawkes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1256aedf66b062c011959c422df1fa08e9f55522">[1256aedf66]</a> Alexander Traud -- chan_sip: Do not send all codecs on INVITE.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25484">ASTERISK-25484</a>: [patch] autoframing=yes has no effect<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=869ef2a8ee4e4df271227d6b9b48470e44ad4831">[869ef2a8ee]</a> Alexander Traud -- chan_sip: Fix autoframing=yes.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25364">ASTERISK-25364</a>: [patch]Issue a TCP connection(kernel) and thread of asterisk is not released<br/>Reported by: Hiroaki Komatsu<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14b41115e363766633aec67f67e9764521b74f5c">[14b41115e3]</a> Jonathan Rose -- chan_sip: Add TCP/TLS keepalive to TCP/TLS server</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25610">ASTERISK-25610</a>: Asterisk crash during "sip reload"<br/>Reported by: Dudás József<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b992014dcf8f1d343e95a06868d4ebd14619d33">[2b992014dc]</a> Richard Mudgett -- chan_sip: Fix crash involving the bogus peer during sip reload.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25476">ASTERISK-25476</a>: chan_sip loses registrations after a while<br/>Reported by: Michael Keuter<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7c88e11aa753e046fe58a19ac82320c81cc6e2b">[e7c88e11aa]</a> Richard Mudgett -- sched.c: Make not return a sched id of 0.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4aed349a7bd2e62d82c5e9535f7cf69263eeb60a">[4aed349a7b]</a> Richard Mudgett -- Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions)</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d9156d10f8941f1da90bf81109904432a2f293d">[6d9156d10f]</a> Richard Mudgett -- Audit improper usage of scheduler exposed by 5c713fdf18f.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07583c288828a496cd7730b55112128fea31eaef">[07583c2888]</a> Steve Davies -- Further fixes to improper usage of scheduler</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24543">ASTERISK-24543</a>: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs<br/>Reported by: Taylor Hawkes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1256aedf66b062c011959c422df1fa08e9f55522">[1256aedf66]</a> Alexander Traud -- chan_sip: Do not send all codecs on INVITE.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><br><h4>Category: Channels/chan_sip/IPv6</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25443">ASTERISK-25443</a>: [patch]IPv6 - Potential issue in via header parsing<br/>Reported by: ffs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f939e2bd48598d721aa18af8182b7b6b91a9fe95">[f939e2bd48]</a> Florian Sauerteig -- chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers.</li>
</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25135">ASTERISK-25135</a>: [patch]RTP Timeout hangup cause code missing<br/>Reported by: Olle Johansson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f8707ae9a57b47a742c051e6714416f46b156118">[f8707ae9a5]</a> Olle Johansson -- channels/chan_sip: Set cause code to 44 on RTP timeout</li>
</ul><br><h4>Category: Channels/chan_sip/T.38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25609">ASTERISK-25609</a>: [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)<br/>Reported by: Filip Jenicek<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=142d4fefb8db0ac2c30b18f75dc415093fb77f27">[142d4fefb8]</a> Filip Jenicek -- chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c)</li>
</ul><br><h4>Category: Channels/chan_sip/WebSocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">ASTERISK-24146</a>: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec<br/>Reported by: Aleksei Kulakov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28d924307940700ce2321572b016fdd8263ac7ad">[28d9243079]</a> Eugene Voityuk -- chan_sip.c: Start ICE negotiation when response is sent or received.</li>
</ul><br><h4>Category: Channels/chan_skinny</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25616">ASTERISK-25616</a>: Warning with a Codec Module which supports PLC with FEC<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=69e3d40ad74883db9bb9b34d6aed71a536e8cf3c">[69e3d40ad7]</a> Alexander Traud -- translate: Avoid a warning message when doing FEC within Opus Codec.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25498">ASTERISK-25498</a>: Asterisk crashes when negotiating g729 without that module installed<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=072d94183ce2b6d2272543732dd5d47390099bb3">[072d94183c]</a> Jonathan Rose -- Fix crash in audiohook translate to slin</li>
</ul><br><h4>Category: Codecs/codec_resample</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25599">ASTERISK-25599</a>: [patch] SLIN Resampling Codec only 80 msec<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=876600ce6e1b165dd068f30c763e5c517c3b6ae8">[876600ce6e]</a> Alexander Traud -- codec_resample: Increase buffer for Opus Codec with FEC.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b76c196e13c08f022f99defc13ec98f0942c2fec">[b76c196e13]</a> Alexander Traud -- codec_resample: Increase buffer for Opus Codec.</li>
</ul><br><h4>Category: Core/AstDB</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25400">ASTERISK-25400</a>: Hints broken when "CustomPresence" doesn't exist in AstDB<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3357678b949ebbc2f7aa00144d19bdbcfb1896b1">[3357678b94]</a> Ivan Poddubny -- func_presencestate: Return "not_set" when no data is set in AstDB</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25600">ASTERISK-25600</a>: bridging: Inconsistency in BRIDGEPEER<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eadad24b595c3b6e5f0472f9936e7e37259308b5">[eadad24b59]</a> Jonathan Rose -- Unset BRIDGEPEER when leaving a bridge</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25434">ASTERISK-25434</a>: Compiler flags not reported in 'core show settings' despite usage during compilation<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d098d00424a3c7ae2c2b2b26ce31d0889c506478">[d098d00424]</a> Corey Farrell -- Fix cli display of build options.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25601">ASTERISK-25601</a>: json: Audit reference usage and thread safety<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=acd19d5f1f84b5c10acc2828b5e73cecfa1ed6ba">[acd19d5f1f]</a> Joshua Colp -- json: Audit ast_json_* usage for thread safety.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25585">ASTERISK-25585</a>: [patch]rasterisk never hits most of main(), but it's assumed to<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b2787876d67cb7d47ce1c7a87db515adcacc151f">[b2787876d6]</a> Walter Doekes -- main: Slight refactor of main. Improve color situation.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25552">ASTERISK-25552</a>: hashtab: Improve NULL tolerance<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afd9a89e5a0ed041d576afa1f387000404ed3c4d">[afd9a89e5a]</a> Joshua Colp -- hashtab: Add NULL check when destroying iterator.</li>
</ul><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=07583c288828a496cd7730b55112128fea31eaef">[07583c2888]</a> Steve Davies -- Further fixes to improper usage of scheduler</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b714b2152d2ee2f6599e9decbe927d4215b6169d">[b714b2152d]</a> Matt Jordan -- res/res_rtp_asterisk: Fix assignment after ao2 decrement</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50fa9ff9972e67899dfc4e7e6766c5977d4aae7a">[50fa9ff997]</a> Matt Jordan -- Fix improper usage of scheduler exposed by 5c713fdf18f</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25546">ASTERISK-25546</a>: threadpool: Race condition between idle timeout and activation<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b818d70533916aa80af7607f225e0b1e98f41648">[b818d70533]</a> Joshua Colp -- threadpool: Handle worker thread transitioning to dead when going active.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-7803">ASTERISK-7803</a>: [patch] Update the maximum packetization values in frame.c<br/>Reported by: dea<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84ff075d411671a801f7de45d7bac48fe4f04a23">[84ff075d41]</a> Alexander Traud -- format: Update the maximum packetization time for iLBC 30.</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25391">ASTERISK-25391</a>: AMI GetConfigJSON returns invalid JSON<br/>Reported by: Bojan Nemčić<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74635b56381c5facaf5b5a6d12a5aa39abf10a0e">[74635b5638]</a> Ivan Poddubny -- manager: Fix GetConfigJSON returning invalid JSON</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=20b22635dcc402e6b1cc39f796ea9e952e0da7ef">[20b22635dc]</a> Kevin Harwell -- pbx: Deadlock between contexts container and context_merge locks</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25625">ASTERISK-25625</a>: res_sorcery_memory_cache: Add full backend caching<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=11ff4e6b3fb6ef9b1470d348e1f43090fbe17a17">[11ff4e6b3f]</a> Joshua Colp -- res_sorcery_memory_cache: Add support for a full backend cache.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25165">ASTERISK-25165</a>: Testsuite - Sorcery memory cache leaks<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc45f4040df77fda402a50822f027f870d114913">[fc45f4040d]</a> Richard Mudgett -- res_sorcery_realtime.c: Fix crash from NULL sorcery object type.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25373">ASTERISK-25373</a>: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d1bdb9d3b1c993b98fdf5041c11708742867820">[6d1bdb9d3b]</a> Walter Doekes -- func_callerid: Document that CALLERID(pres) is available.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25527">ASTERISK-25527</a>: Quirky xmldoc description wrapping<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d425f2eb439161065b43d1178bfdac632d8bf56">[0d425f2eb4]</a> Walter Doekes -- xmldoc: Improve xmldoc wrapping of 'core show ...' output.</li>
</ul><br><h4>Category: Formats/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25664">ASTERISK-25664</a>: ast_format_cap_append_by_type leaks a reference<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0766192ee4762c564b4f88da640e4dde7d5c987f">[0766192ee4]</a> Corey Farrell -- ast_format_cap_append_by_type: Resolve codec reference leak.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25584">ASTERISK-25584</a>: [patch] format-attribute module: VP8 missing<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a18193dc09e90102f664a94fc9eeea5cac44b71">[5a18193dc0]</a> Alexander Traud -- res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25545">ASTERISK-25545</a>: [patch] translation module gets cached not joint format<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b508789ab225f57e22bce93e243e9d642a73191">[0b508789ab]</a> Alexander Traud -- translate: Provide translation modules the result of SDP negotiation.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25535">ASTERISK-25535</a>: [patch] format creation on module load instead of cache<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e5bf12b33de8db9c53f571e4c4d5cb094a0d008">[4e5bf12b33]</a> Joshua Colp -- format_cap: Don't append the 'none' format when appending all.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f3ac4d8090207dd4440bf279e1d5ce4702aee314">[f3ac4d8090]</a> Alexander Traud -- ast_format_cap: Avoid format creation on module load, use cache instead.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25537">ASTERISK-25537</a>: [patch] format-attribute module: RFC or internal defaults?<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4bf84459c77227b7adc642d04b9ad93659d96ee2">[4bf84459c7]</a> Alexander Traud -- rtp_engine: Init a format-attribute module to its RFC defaults.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25533">ASTERISK-25533</a>: [patch] buffer for ast_format_cap_get_names only 64 bytes<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bff400df7ff1fda353fab49de2fcf9cbba5cd89">[1bff400df7]</a> Alexander Traud -- ast_format_cap_get_names: To display all formats, the buffer was increased.</li>
</ul><br><h4>Category: Formats/format_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25573">ASTERISK-25573</a>: [patch] H.264 format attribute module: resets whole SDP<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1aa552b2a28d51ef9d6ac4f236ee9852b0ca449a">[1aa552b2a2]</a> Alexander Traud -- res_format_attr_h264: Do not reset string buffer.</li>
</ul><br><h4>Category: Functions/func_callerid</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25373">ASTERISK-25373</a>: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d1bdb9d3b1c993b98fdf5041c11708742867820">[6d1bdb9d3b]</a> Walter Doekes -- func_callerid: Document that CALLERID(pres) is available.</li>
</ul><br><h4>Category: Resources/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25584">ASTERISK-25584</a>: [patch] format-attribute module: VP8 missing<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a18193dc09e90102f664a94fc9eeea5cac44b71">[5a18193dc0]</a> Alexander Traud -- res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25441">ASTERISK-25441</a>: Deadlock in res_sorcery_memory_cache.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40c69e78f571781b67a240554c119b870e3cd6d4">[40c69e78f5]</a> Richard Mudgett -- res_sorcery_memory_cache.c: Fix deadlock with scheduler.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dfeb513e85d13550d81b40df5e95333c1ad5c61c">[dfeb513e85]</a> Richard Mudgett -- res_sorcery_memory_cache.c: Replace inline code with function.</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25593">ASTERISK-25593</a>: fastagi: record file closed after sending result<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45efbf8503a29d298a9cb6c5de4925037a642b35">[45efbf8503]</a> Kevin Harwell -- fastagi: record file closed after sending result</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25522">ASTERISK-25522</a>: ARI: Crash when creating channel via ARI originate with requesting channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=506aea26e6c67cd53874aa3ffef278524dfd7878">[506aea26e6]</a> Matt Jordan -- main/dial: Protect access to the format_cap structure of the requesting channel</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25522">ASTERISK-25522</a>: ARI: Crash when creating channel via ARI originate with requesting channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=506aea26e6c67cd53874aa3ffef278524dfd7878">[506aea26e6]</a> Matt Jordan -- main/dial: Protect access to the format_cap structure of the requesting channel</li>
</ul><br><h4>Category: Resources/res_config_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><br><h4>Category: Resources/res_format_attr_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25583">ASTERISK-25583</a>: [patch] format-attribute module: RFC 7587 (Opus Codec)<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e2178c05e5e324482f1fb46e488e96e574284cd">[3e2178c05e]</a> Alexander Traud -- res_format_attr_opus: Update to latest RFC 7587.</li>
</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24106">ASTERISK-24106</a>: WebSockets Automatically decides what driver it will use <br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0393bd6bed3b87b312d7fc252c4fa3782df8260a">[0393bd6bed]</a> Corey Farrell -- chan_sip: Allow websockets to be disabled.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25689">ASTERISK-25689</a>: pjsip show contacts not working in Asterisk 13.7rc2<br/>Reported by: Marcelo Terres<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c5fbd08b81281e7eb7c1401541f82e426c49136">[6c5fbd08b8]</a> Mark Michelson -- res_sorcery_realtime: Remove leading ^ requirement.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25608">ASTERISK-25608</a>: res_pjsip/contacts/statsd: Lifecycle events aren't consistent<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=450579e908279e664cb4364a2e7dd1cfd6a90396">[450579e908]</a> gtjoseph -- res_pjsip/contacts/statsd: Make contact lifecycle events more consistent</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25595">ASTERISK-25595</a>: Unescaped : in messge sent to statsd<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9184fbeb347d1168add1f3140af3b6837c8d78db">[9184fbeb34]</a> gtjoseph -- res_pjsip: Use a MD5 hash for static Contact IDs</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25598">ASTERISK-25598</a>: res_pjsip: Contact status messages are printing a hash instead of the uri<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed9134282e22c6985ce853f53d7569aa5b93ebe0">[ed9134282e]</a> gtjoseph -- res_pjsip: Update logging to show contact->uri in messages</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25486">ASTERISK-25486</a>: res_pjsip: Fix deadlock when validating URIs<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f2725c8b77f6e6d6b70c12c4e57e26083530c3be">[f2725c8b77]</a> Joshua Colp -- res_pjsip: Move URI validation to use time.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><br><h4>Category: Resources/res_pjsip_notify</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25590">ASTERISK-25590</a>: CLI Usage info for 'pjsip send notify' references incorrect config<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b75f587d159cb68ed24b6ee1007ed062f669d79f">[b75f587d15]</a> Corey Farrell -- res_pjsip_notify: Fix CLI usage info</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25575">ASTERISK-25575</a>: res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f71263e72268bb4966fa7d8f68a0a8b99419ec5">[8f71263e72]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Apply configuration on object type load</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25485">ASTERISK-25485</a>: res_pjsip_outbound_registration: registration stops due to 400 response<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c58091737da86e17cbad4d86ebf4f04055e505fa">[c58091737d]</a> Kevin Harwell -- res_pjsip_outbound_registration: registration stops due to fatal 4xx response</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25513">ASTERISK-25513</a>: Crash: malloc failed with high load of subscriptions.<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6fbffe42e13d82eebd5545de9a74b6a36bd9a558">[6fbffe42e1]</a> Mark Michelson -- res_pjsip: Set threadpool max size default to 50.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25505">ASTERISK-25505</a>: res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a021a42adaee95d115aa3200467943fecd1f13a">[9a021a42ad]</a> Joshua Colp -- res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.</li>
</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25582">ASTERISK-25582</a>: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6614babea27fbafbe11820ea03737dd5c4f9ecec">[6614babea2]</a> Matt Jordan -- bridges/bridge_t38: Add a bridging module for managing T.38 state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4875e5ac32f5ccad51add6a4216947bfb385245d">[4875e5ac32]</a> Matt Jordan -- chan_pjsip: Handle T.38 faxes with direct media bridges</li>
</ul><br><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24106">ASTERISK-24106</a>: WebSockets Automatically decides what driver it will use <br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0393bd6bed3b87b312d7fc252c4fa3782df8260a">[0393bd6bed]</a> Corey Farrell -- chan_sip: Allow websockets to be disabled.</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">ASTERISK-24146</a>: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec<br/>Reported by: Aleksei Kulakov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28d924307940700ce2321572b016fdd8263ac7ad">[28d9243079]</a> Eugene Voityuk -- chan_sip.c: Start ICE negotiation when response is sent or received.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25451">ASTERISK-25451</a>: Broken video - erased rtp marker bit<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1435aa3fad5bda73a66dbccf3982787eff55ea2">[a1435aa3fa]</a> Stefan Engström -- res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending</li>
</ul><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=56ed7b9dd560e468be31684e56a8070b88ae0205">[56ed7b9dd5]</a> Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug.</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25595">ASTERISK-25595</a>: Unescaped : in messge sent to statsd<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9184fbeb347d1168add1f3140af3b6837c8d78db">[9184fbeb34]</a> gtjoseph -- res_pjsip: Use a MD5 hash for static Contact IDs</li>
</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25582">ASTERISK-25582</a>: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6614babea27fbafbe11820ea03737dd5c4f9ecec">[6614babea2]</a> Matt Jordan -- bridges/bridge_t38: Add a bridging module for managing T.38 state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4875e5ac32f5ccad51add6a4216947bfb385245d">[4875e5ac32]</a> Matt Jordan -- chan_pjsip: Handle T.38 faxes with direct media bridges</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25165">ASTERISK-25165</a>: Testsuite - Sorcery memory cache leaks<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc45f4040df77fda402a50822f027f870d114913">[fc45f4040d]</a> Richard Mudgett -- res_sorcery_realtime.c: Fix crash from NULL sorcery object type.</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25615">ASTERISK-25615</a>: res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36097a185db00230a89f019b9b8ee2d478cc6665">[36097a185d]</a> Richard Mudgett -- Fix sscanf() format string type mismatch.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b867fa9043dec7aee8fbe21a6537efb103e4d92">[5b867fa904]</a> gtjoseph -- pjsip/config_transport: Check pjproject version at runtime for async ops</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e03582a1c293c0ed7e37896758be613e3e281bfd">[e03582a1c2]</a> gtjoseph -- res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25513">ASTERISK-25513</a>: Crash: malloc failed with high load of subscriptions.<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6fbffe42e13d82eebd5545de9a74b6a36bd9a558">[6fbffe42e1]</a> Mark Michelson -- res_pjsip: Set threadpool max size default to 50.</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24718">ASTERISK-24718</a>: [patch]Add inital support of "sanitize" to configure<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=486b172b50ae5b525d03ea7467bdb4ffa7ad90fd">[486b172b50]</a> Ivan Poddubny -- Build: Add menuselect options for using compiler sanitizers</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25518">ASTERISK-25518</a>: taskprocessor: Add high water mark<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ff48319d9a3d0e4dd301f90d4b9b214f9f87e3a">[6ff48319d9]</a> Jonathan Rose -- taskprocessor: Add high water mark warnings</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25618">ASTERISK-25618</a>: res_pjsip: Check for readability of TLS files at startup<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21962dad93fdb887899676597779a6ae47ff1edb">[21962dad93]</a> gtjoseph -- res_pjsip: Add existence and readablity checks for tls related files</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25477">ASTERISK-25477</a>: pjsip show "command" like [criteria]<br/>Reported by: Bryant Zimmerman<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=162acd45f744395c19ec5686af30d0abd61ef897">[162acd45f7]</a> gtjoseph -- res_pjsip: Add "like" processing to pjsip list and show commands</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25572">ASTERISK-25572</a>: Endpoints: Add StatsD stats for Asterisk endpoints<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d27aac0a9d4b7b72ddb73ae45f6f7327110a07dc">[d27aac0a9d]</a> Matt Jordan -- res/res_endpoint_stats: Add module to emit endpoint StatsD statistics</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</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: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25645">ASTERISK-25645</a>: res_rtp_asterisk: Lock inversion<br/>Reported by: Steve Davies<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21a9bc1b812ad15ee911e6aa7c37385027b18f15">[21a9bc1b81]</a> Joshua Colp -- res_rtp_asterisk: Revert DTLS negotiation changes.</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=e42707cf4a35d8d5349021ba6699070f8304b10f">e42707cf4a</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=e66f043868f225bd056dd89f785568d56b2dba06">e66f043868</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3f6147748f448d201dd940d5a54db0d343f3264">c3f6147748</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3086c36c8d4c3e2d9c093494d4879ae06194518b">3086c36c8d</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27f82f20306c76c7da11a1ac78cbd9923615594d">27f82f2030</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.7.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5cb9e3f1d559e3e4ac9c8a866fe10464fc862f7">a5cb9e3f1d</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.7.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8043875fa1df063192999c1c7a489ae0b490cfc6">8043875fa1</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=a0e55b01c4f77bd72a36d13e2dd2cbdbd2c206d9">a0e55b01c4</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae3b9d0958d83acbfd8e3ab374b5bf36b2a67e1b">ae3b9d0958</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ebc1646def2bcd0bf65a73756b62ffd420dccef1">ebc1646def</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7792775db142f515b2701e41ee25ae699ad57e4f">7792775db1</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.7.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a93a5387d4242c501b434c80f43a6b09c729dc38">a93a5387d4</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.7.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e201b742a804ff003d86e7c8566856210d02026">8e201b742a</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=5a164c70f2dadcdcf325ccf326efc6edc2315d1b">5a164c70f2</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e039eca0a7ee4589b64cc8dbd0d18700eed7a31f">e039eca0a7</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bfe2eb875168864d243dfbed2d4b184e72001110">bfe2eb8751</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=805297783dc96976c5806d371f1618a5a3901ff4">805297783d</a></td><td>Mark Michelson</td><td>Alembic: Add PJSIP global keep_alive_interval.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=63df9bb56001a29765a7e85702019144380581cf">63df9bb560</a></td><td>Mark Michelson</td><td>Alembic: Increase column size of PJSIP AOR "contact".</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9bc1e49325c0319418c1bbd8ee8e04f31946ef8c">9bc1e49325</a></td><td>Joshua Colp</td><td>rtp_engine: Ignore empty filenames in DTLS configuration.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c78eb1e82bc6b92ced080df5766b3fbd7dedeb72">c78eb1e82b</a></td><td>Joshua Colp</td><td>chan_sip: Enable WebSocket support by default.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4efe7bf05179c25b96c6a0dea401d4847c7737af">4efe7bf051</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0370acecfc4095c1bc89f692d642e2b944387f0a">0370acecfc</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1bb33fe0b4e655e7ed7f23710bc695ef88f3640">d1bb33fe0b</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d06a65de0111e34df6d0a1a304bab787dd705094">d06a65de01</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb37b44660983ae85680f788a0f1e9752e94cad0">fb37b44660</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20b7164b8c68f99b0fda7b9951b975046261af99">20b7164b8c</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cbf2414c365cd3bafb20552c2630fc10077170d">6cbf2414c3</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ba1794464dbbd15089a317494f31bf5c1a259ff7">ba1794464d</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3e9753a234049dde8ddc9ea2df561142a1f0e3f">b3e9753a23</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b0df64b5f0665fad4abfe75792241a5b5b187bd1">b0df64b5f0</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ce9a59faf6aa584644434c1f9b0d7ae1145b016b">ce9a59faf6</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2e26bef5bbf1966c4d3b29348c53b5fe529f6c28">2e26bef5bb</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9b47516d6544a179a8f8944e8bc612ead34591">5e9b47516d</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=034112c5747e51601097dc90de420923624559a8">034112c574</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1f8ff1789ccf9fc226d579400f9d3c38bfb0bf5">d1f8ff1789</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9376488bef3f2cd377030b7890dbb8cc05efbce9">9376488bef</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a894c9e7a919b3c17b8798d465435164918aa27e">a894c9e7a9</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52afb0f112cf23cd306775616efb42551d80131d">52afb0f112</a></td><td>Kevin Harwell</td><td>.version: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2de343eb85227127b3b447490142adedd5afdc02">2de343eb85</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=184de2a16086e3677bbf84a5b08f67192aa1fcb0">184de2a160</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.7.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94f9927784e63d0e30aa7919b83b0e0fcc35c57e">94f9927784</a></td><td>Matt Jordan</td><td>main/utils: Don't emit an ERROR message if the read end of a pipe closes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=529535f0c2dd2fca84e31287dd7a00622cacd3c8">529535f0c2</a></td><td>Matt Jordan</td><td>Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb0b60619daddb1448e980f9067780bcd6ca5e35">bb0b60619d</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Fix off nominal ref leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3fcf160faeb036529c575b66d73e7978f475fb28">3fcf160fae</a></td><td>Niklas Larsson</td><td>CHANGES: Fix a typo</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59881fbb9988be2f4e07ca750f45a404e79cb115">59881fbb99</a></td><td>David M. Lee</td><td>Fixed some typos</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b94d9a10d5001ddb2c6a9aee4b66ee92ec3a3c8">2b94d9a10d</a></td><td>Matt Jordan</td><td>res/res_pjsip_t38: Add debug statements</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af288b2d9650bf7fdc30591e82a06b6c7610b80f">af288b2d96</a></td><td>Matt Jordan</td><td>main/cli: Use proper string methods to check existence of context/exten/app</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3354b325c67824b4aa052fb81693d28e792886a6">3354b325c6</a></td><td>Matt Jordan</td><td>res_statsd: Add functions that support variable arguments</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4a522d587bb1986cc66ed59a087be3784eaaceb">d4a522d587</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e44ab3816cae3e85d27e969366a12881af42fa46">e44ab3816c</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Fix 423 response handling.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f62b642fe32d06ca24b4a0de94543279fe918d0a">f62b642fe3</a></td><td>Matt Jordan</td><td>res/res_pjsip: Fix off nominal crash with requests that fail and have a timer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0f2f8de458e68e412be91ccc363a9f7aae77c78">c0f2f8de45</a></td><td>Richard Mudgett</td><td>res_pjsip_rfc3326.c: Fix crash when channel goes away.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f43b85c92050c8deba7041e687404228294d920">4f43b85c92</a></td><td>Mark Michelson</td><td>Taskprocessors: Increase high-water mark</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=367972e42d1a5f73d8bb4abacd4c681fc77dcd24">367972e42d</a></td><td>Mark Michelson</td><td>res_pjsip distributor: Don't send 503 response to responses.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f9cb7d62bf1ee2d3f7d878607d2d1eb9995dd03">2f9cb7d62b</a></td><td>Mark Michelson</td><td>res_pjsip: Deny requests when threadpool queue is backed up.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8410336681b54766f148b7708f08d6d5e7ff6f2e">8410336681</a></td><td>Walter Doekes</td><td>docs: Fix a few typo's in app docs (more then, resourse).</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afec1b1b6497c2b81c0ef468861933b6ba277562">afec1b1b64</a></td><td>Matt Jordan</td><td>res_pjsip/location: Destroy contact_status objects on contact deletion</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=715f770c9ff011284c1e87f9b5bcde1fc02ab4df">715f770c9f</a></td><td>Matt Jordan</td><td>pjsip_configuration: On delete, remove the persistent version of an endpoint</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0f190af08c5f7f4af3316209b174bd92be145c3">f0f190af08</a></td><td>Matt Jordan</td><td>main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=11e54b1932b62a084722cb547a51a5fc2ca4d423">11e54b1932</a></td><td>Matt Jordan</td><td>pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=118d628e084311af0096178b096e3959ac603edd">118d628e08</a></td><td>Matt Jordan</td><td>Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ebe69dee0d0132bcda93ce909d0a82316e86e8f7">ebe69dee0d</a></td><td>Mark Michelson</td><td>format_cap: Detect vector allocation failures.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b19efefef0cdc7203611bf9d161766ef6922558">3b19efefef</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a346f095fb5f02391d870d2527a665ea926e65b">0a346f095f</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Ensure dialog lock balance.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad395080957b33a17f6cfe2c83697bebef286c25">ad39508095</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Prevent crashes on final NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=067f40876001255aed9bf8b65567d1c25961aebd">067f408760</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Remove serializer when sending final NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bcc5927655c71f2ea0db31c0cd0a3bf0095714d">1bcc592765</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3cc2bd7dfad379cec77e7333cc93c23fda6aa92">b3cc2bd7df</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Solidify lifetime and ownership of objects.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8c65dfa413cff6ad5af12350564f37d4786fe01">c8c65dfa41</a></td><td>Richard Mudgett</td><td>strings.c: Fix __ast_str_helper() to always return a terminated string.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b271d4a28a607341f2374b6f8200b7f4f775e5e6">b271d4a28a</a></td><td>Richard Mudgett</td><td>Add missing failure checks to ast_str_set_va() callers.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fd2adc20430221dadd58c60d57655b95da168c6">9fd2adc204</a></td><td>Matt Jordan</td><td>rest-api-templates: Wikify error code response reasons</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fc9777fa34753fb38991d42d8dbed516e907ca2">9fc9777fa3</a></td><td>Matt Jordan</td><td>contrib/scripts/autosupport: Update for Asterisk 13</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e14023ca3543d91bc108f8f21af0509c2a428e47">e14023ca35</a></td><td>Richard Mudgett</td><td>config.c: Fix off-nominal memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a99e821520405be48241f2a51c659cefd6939da2">a99e821520</a></td><td>Richard Mudgett</td><td>config.c: Fix potential memory corruption after [section](+).</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f777ab58499f6b3b9264c9bd6750e0ad5eb033c">8f777ab584</a></td><td>Debian Amtelco</td><td>chan_pjsip: Add Referred-By header to the PJSIP REFER packet.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ced0a2d71b690f24026392bcfbbe4c36eb8d4dff">ced0a2d71b</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Shutdown in a less crash potential order.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc279eea11853ad90605775a63d58a1cab88f96c">cc279eea11</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Misc tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9af3b613f6423e73a28546df5808155a9fc3cfa3">9af3b613f6</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddebb217f00c00e95259d3977d6d8b43adc69c65">ddebb217f0</a></td><td>Richard Mudgett</td><td>sched.c: Add warning about negative time interval request.</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-summary.html | 322 -
asterisk-13.6.0-summary.txt | 839 ---
b/.version | 2
b/CHANGES | 90
b/ChangeLog | 2138 +++++++++-
b/Makefile | 64
b/Makefile.rules | 20
b/addons/chan_mobile.c | 2
b/addons/chan_ooh323.c | 14
b/apps/app_chanisavail.c | 6
b/apps/app_confbridge.c | 18
b/apps/app_dial.c | 8
b/apps/app_dumpchan.c | 2
b/apps/app_meetme.c | 3
b/apps/app_page.c | 2
b/apps/app_queue.c | 1
b/apps/confbridge/conf_config_parser.c | 4
b/apps/confbridge/include/confbridge.h | 2
b/bridges/bridge_native_rtp.c | 4
b/build_tools/cflags.xml | 28
b/build_tools/make_version_c | 25
b/channels/chan_alsa.c | 2
b/channels/chan_console.c | 2
b/channels/chan_dahdi.c | 5
b/channels/chan_iax2.c | 65
b/channels/chan_mgcp.c | 10
b/channels/chan_motif.c | 2
b/channels/chan_nbs.c | 2
b/channels/chan_oss.c | 2
b/channels/chan_phone.c | 2
b/channels/chan_pjsip.c | 32
b/channels/chan_sip.c | 165
b/channels/chan_skinny.c | 53
b/channels/chan_unistim.c | 18
b/channels/chan_vpb.cc | 2
b/channels/sig_pri.c | 2
b/channels/sip/config_parser.c | 9
b/channels/sip/include/sip.h | 1
b/channels/sip/reqresp_parser.c | 2
b/codecs/codec_resample.c | 8
b/configs/samples/confbridge.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 8
b/configs/samples/sip.conf.sample | 4
b/contrib/ast-db-manage/config/versions/189a235b3fd7_add_keep_alive_interval.py | 22
b/contrib/ast-db-manage/config/versions/28ce1e718f05_add_fatal_response_interval.py | 22
b/contrib/ast-db-manage/config/versions/2d078ec071b7_increaes_contact_column_size.py | 22
b/contrib/realtime/mssql/mssql_config.sql | 20
b/contrib/realtime/mysql/mysql_config.sql | 18
b/contrib/realtime/oracle/oracle_config.sql | 20
b/contrib/realtime/postgresql/postgresql_config.sql | 14
b/contrib/scripts/autosupport | 95
b/funcs/func_callerid.c | 39
b/funcs/func_channel.c | 4
b/funcs/func_holdintercept.c | 236 +
b/funcs/func_presencestate.c | 4
b/include/asterisk/ast_version.h | 3
b/include/asterisk/bridge.h | 12
b/include/asterisk/format_cap.h | 5
b/include/asterisk/res_pjsip.h | 17
b/include/asterisk/res_pjsip_cli.h | 2
b/include/asterisk/res_pjsip_pubsub.h | 9
b/include/asterisk/statsd.h | 71
b/include/asterisk/taskprocessor.h | 8
b/include/asterisk/term.h | 4
b/include/asterisk/threadpool.h | 6
b/include/asterisk/translate.h | 8
b/include/asterisk/utils.h | 23
b/main/aoc.c | 20
b/main/asterisk.c | 124
b/main/audiohook.c | 15
b/main/bridge.c | 10
b/main/bridge_channel.c | 4
b/main/channel.c | 13
b/main/cli.c | 8
b/main/codec_builtin.c | 2
b/main/config.c | 44
b/main/dial.c | 7
b/main/file.c | 2
b/main/format.c | 22
b/main/format_cap.c | 47
b/main/hashtab.c | 2
b/main/loader.c | 4
b/main/manager.c | 22
b/main/pbx.c | 595 +-
b/main/rtp_engine.c | 38
b/main/sched.c | 22
b/main/sorcery.c | 8
b/main/stasis.c | 4
b/main/stasis_channels.c | 24
b/main/stasis_endpoints.c | 2
b/main/strings.c | 91
b/main/taskprocessor.c | 18
b/main/term.c | 27
b/main/threadpool.c | 72
b/main/translate.c | 41
b/main/utils.c | 51
b/main/xmldoc.c | 170
b/pbx/pbx_dundi.c | 1
b/res/res_agi.c | 6
b/res/res_chan_stats.c | 4
b/res/res_config_pgsql.c | 8
b/res/res_endpoint_stats.c | 157
b/res/res_fax.c | 4
b/res/res_format_attr_h264.c | 16
b/res/res_format_attr_opus.c | 210
b/res/res_format_attr_vp8.c | 228 +
b/res/res_pjsip.c | 63
b/res/res_pjsip/config_auth.c | 15
b/res/res_pjsip/config_system.c | 2
b/res/res_pjsip/config_transport.c | 50
b/res/res_pjsip/location.c | 147
b/res/res_pjsip/pjsip_cli.c | 15
b/res/res_pjsip/pjsip_configuration.c | 270 -
b/res/res_pjsip/pjsip_distributor.c | 16
b/res/res_pjsip/pjsip_options.c | 129
b/res/res_pjsip_caller_id.c | 14
b/res/res_pjsip_endpoint_identifier_ip.c | 58
b/res/res_pjsip_exten_state.c | 4
b/res/res_pjsip_mwi.c | 6
b/res/res_pjsip_notify.c | 2
b/res/res_pjsip_outbound_registration.c | 125
b/res/res_pjsip_pubsub.c | 338 +
b/res/res_pjsip_pubsub.exports.in | 1
b/res/res_pjsip_rfc3326.c | 17
b/res/res_pjsip_sdp_rtp.c | 11
b/res/res_pjsip_t38.c | 67
b/res/res_rtp_asterisk.c | 6
b/res/res_sorcery_memory_cache.c | 1143 ++++-
b/res/res_sorcery_realtime.c | 10
b/res/res_stasis.c | 6
b/res/res_stasis_playback.c | 4
b/res/res_stasis_recording.c | 4
b/res/res_statsd.c | 88
b/res/res_statsd.exports.in | 1
b/res/stasis/app.c | 4
b/rest-api-templates/api.wiki.mustache | 2
b/rest-api-templates/asterisk_processor.py | 2
b/tests/test_config.c | 4
b/tests/test_format_cap.c | 4
b/tests/test_sorcery_realtime.c | 10
140 files changed, 6674 insertions(+), 2787 deletions(-)</pre><br></html>

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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-13.7.2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.7.2</h3><h3 align="center">Date: 2016-02-05</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-13.7.1.</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 Mark Michelson <mmichelson@lunkwill><br/>2 Mark Michelson <mmichelson@lunkwill.digium.internal><br/>1 Alexei Gradinari License #5691<br/></td><td width="33%"><td width="33%">1 Nic Colledge <nic@njcolledge.net><br/>1 Nic Colledge<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: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8de94229baa3ee8471521c7c0c59424de0131c61">[8de94229ba]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8de94229baa3ee8471521c7c0c59424de0131c61">[8de94229ba]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8de94229baa3ee8471521c7c0c59424de0131c61">[8de94229ba]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</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=0ba44bd6f3b2db32272c43c70d6c265d3213a033">0ba44bd6f3</a></td><td>Mark Michelson</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b750dfe202ce8153a7d56f99df9561a8ad9e97ea">b750dfe202</a></td><td>Mark Michelson</td><td>.version: Update for 13.7.2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1b94ffe78e8240e374c2693449a6497eb0beb80">c1b94ffe78</a></td><td>Mark Michelson</td><td>.lastclean: Update for 13.7.2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=932ed1ab5bcbecce3d19da579a5229c0308e87fa">932ed1ab5b</a></td><td>Mark Michelson</td><td>realtime: Add database scripts for 13.7.2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a56f55d566023cd43bdfb80982e454e795878a1c">a56f55d566</a></td><td>Mark Michelson</td><td>Check for OpenSSL defines before trying to use them.</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-13.7.1-summary.html | 34 ---------
asterisk-13.7.1-summary.txt | 155 -------------------------------------------
b/.version | 2
b/configure | 89 ++++++++++++++++++++++++
4 files changed, 90 insertions(+), 190 deletions(-)</pre><br></html>

131
asterisk-13.7.2-summary.txt Normal file
View File

@@ -0,0 +1,131 @@
Release Summary
asterisk-13.7.2
Date: 2016-02-05
<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-13.7.1.
----------------------------------------------------------------------
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 Mark Michelson 1 Nic Colledge
2 Mark Michelson 1 Nic Colledge
1 Alexei Gradinari License #5691
----------------------------------------------------------------------
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: Channels/chan_pjsip
ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to
asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic Colledge
* [8de94229ba] Alexei Gradinari License #5691 -- res_sorcery_realtime:
Fix regex regression.
Category: Core/Sorcery
ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to
asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic Colledge
* [8de94229ba] Alexei Gradinari License #5691 -- res_sorcery_realtime:
Fix regex regression.
Category: Resources/res_pjsip
ASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to
asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic Colledge
* [8de94229ba] Alexei Gradinari License #5691 -- res_sorcery_realtime:
Fix regex regression.
----------------------------------------------------------------------
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 |
|------------+----------------+------------------------------------------|
| 0ba44bd6f3 | Mark Michelson | Release summaries: Remove previous |
| | | versions |
|------------+----------------+------------------------------------------|
| b750dfe202 | Mark Michelson | .version: Update for 13.7.2 |
|------------+----------------+------------------------------------------|
| c1b94ffe78 | Mark Michelson | .lastclean: Update for 13.7.2 |
|------------+----------------+------------------------------------------|
| 932ed1ab5b | Mark Michelson | realtime: Add database scripts for |
| | | 13.7.2 |
|------------+----------------+------------------------------------------|
| a56f55d566 | Mark Michelson | Check for OpenSSL defines before trying |
| | | to use them. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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-13.7.1-summary.html | 34 ---------
asterisk-13.7.1-summary.txt | 155 -------------------------------------------
b/.version | 2
b/configure | 89 ++++++++++++++++++++++++
4 files changed, 90 insertions(+), 190 deletions(-)

View File

@@ -3965,6 +3965,13 @@ static int retrans_pkt(const void *data)
}
/* For non-invites, a maximum of 4 secs */
if (INT_MAX / pkt->timer_a < pkt->timer_t1) {
/*
* Uh Oh, we will have an integer overflow.
* Recalculate previous timeout time instead.
*/
pkt->timer_a = pkt->timer_a / 2;
}
siptimer_a = pkt->timer_t1 * pkt->timer_a; /* Double each time */
if (pkt->method != SIP_INVITE && siptimer_a > 4000) {
siptimer_a = 4000;

View File

@@ -90,6 +90,26 @@ bindaddr=127.0.0.1
; private in same .pem file.
; openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
;
; tlscipher= ; The list of allowed ciphers
; ; if none are specified the following cipher
; ; list will be used instead:
; ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:
; ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:
; kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:
; ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:
; ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:
; DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:
; AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:
; AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:
; !EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA
;
; tlsdisablev1=yes ; Disable TLSv1 support - if not set this defaults to "yes"
; tlsdisablev11=yes ; Disable TLSv1.1 support - if not set this defaults to "no"
; tlsdisablev12=yes ; Disable TLSv1.2 support - if not set this defaults to "no"
;
; tlsservercipherorder=yes ; Use the server preference order instead of the client order
; ; Defaults to "yes"
;
; The post_mappings section maps URLs to real paths on the filesystem. If a
; POST is done from within an authenticated manager session to one of the
; configured POST mappings, then any files in the POST will be placed in the

98
configure vendored
View File

@@ -651,6 +651,8 @@ PBX_MSG_NOSIGNAL
PBX_IXJUSER
GMIME_LIBS
GMIME_CFLAGS
PBX_SSL_OP_NO_TLSV1_2
PBX_SSL_OP_NO_TLSV1_1
OPENH323_BUILD
OPENH323_SUFFIX
OPENH323_LIBDIR
@@ -30329,6 +30331,102 @@ rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
fi
if test "$PBX_OPENSSL" = "1";
then
if test "x${PBX_SSL_OP_NO_TLSV1_1}" != "x1"; then
{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for SSL_OP_NO_TLSv1_1 in openssl/ssl.h" >&5
$as_echo_n "checking for SSL_OP_NO_TLSv1_1 in openssl/ssl.h... " >&6; }
saved_cppflags="${CPPFLAGS}"
if test "x${SSL_OP_NO_TLSV1_1_DIR}" != "x"; then
SSL_OP_NO_TLSV1_1_INCLUDE="-I${SSL_OP_NO_TLSV1_1_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${SSL_OP_NO_TLSV1_1_INCLUDE}"
cat confdefs.h - <<_ACEOF >conftest.$ac_ext
/* end confdefs.h. */
#include <openssl/ssl.h>
int
main ()
{
#if defined(SSL_OP_NO_TLSv1_1)
int foo = 0;
#else
int foo = bar;
#endif
0
;
return 0;
}
_ACEOF
if ac_fn_c_try_compile "$LINENO"; then :
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
$as_echo "yes" >&6; }
PBX_SSL_OP_NO_TLSV1_1=1
$as_echo "#define HAVE_SSL_OP_NO_TLSV1_1 1" >>confdefs.h
else
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
$as_echo "no" >&6; }
fi
rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
CPPFLAGS="${saved_cppflags}"
fi
if test "x${PBX_SSL_OP_NO_TLSV1_2}" != "x1"; then
{ $as_echo "$as_me:${as_lineno-$LINENO}: checking for SSL_OP_NO_TLSv1_2 in openssl/ssl.h" >&5
$as_echo_n "checking for SSL_OP_NO_TLSv1_2 in openssl/ssl.h... " >&6; }
saved_cppflags="${CPPFLAGS}"
if test "x${SSL_OP_NO_TLSV1_2_DIR}" != "x"; then
SSL_OP_NO_TLSV1_2_INCLUDE="-I${SSL_OP_NO_TLSV1_2_DIR}/include"
fi
CPPFLAGS="${CPPFLAGS} ${SSL_OP_NO_TLSV1_2_INCLUDE}"
cat confdefs.h - <<_ACEOF >conftest.$ac_ext
/* end confdefs.h. */
#include <openssl/ssl.h>
int
main ()
{
#if defined(SSL_OP_NO_TLSv1_2)
int foo = 0;
#else
int foo = bar;
#endif
0
;
return 0;
}
_ACEOF
if ac_fn_c_try_compile "$LINENO"; then :
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: yes" >&5
$as_echo "yes" >&6; }
PBX_SSL_OP_NO_TLSV1_2=1
$as_echo "#define HAVE_SSL_OP_NO_TLSV1_2 1" >>confdefs.h
else
{ $as_echo "$as_me:${as_lineno-$LINENO}: result: no" >&5
$as_echo "no" >&6; }
fi
rm -f core conftest.err conftest.$ac_objext conftest.$ac_ext
CPPFLAGS="${saved_cppflags}"
fi
fi
if test "x${PBX_SRTP}" != "x1" -a "${USE_SRTP}" != "no"; then
pbxlibdir=""

View File

@@ -2289,6 +2289,12 @@ then
AST_C_DECLARE_CHECK([OPENSSL_ECDH_AUTO], [SSL_CTX_set_ecdh_auto], [openssl/ssl.h])
fi
if test "$PBX_OPENSSL" = "1";
then
AST_C_DEFINE_CHECK([SSL_OP_NO_TLSV1_1], [SSL_OP_NO_TLSv1_1], [openssl/ssl.h])
AST_C_DEFINE_CHECK([SSL_OP_NO_TLSV1_2], [SSL_OP_NO_TLSv1_2], [openssl/ssl.h])
fi
AST_EXT_LIB_CHECK([SRTP], [srtp], [srtp_init], [srtp/srtp.h])
if test "$PBX_SRTP" = "1";

View File

@@ -826,6 +826,12 @@
/* Define to 1 if you have the ISDN SS7 library. */
#undef HAVE_SS7
/* Define if your system has the SSL_OP_NO_TLSV1_1 headers. */
#undef HAVE_SSL_OP_NO_TLSV1_1
/* Define if your system has the SSL_OP_NO_TLSV1_2 headers. */
#undef HAVE_SSL_OP_NO_TLSV1_2
/* Define to 1 if `stat' has the bug that it succeeds when given the
zero-length file name argument. */
#undef HAVE_STAT_EMPTY_STRING_BUG

View File

@@ -86,7 +86,15 @@ enum ast_ssl_flags {
/*! Use SSLv3 for outgoing client connections */
AST_SSL_SSLV3_CLIENT = (1 << 4),
/*! Use TLSv1 for outgoing client connections */
AST_SSL_TLSV1_CLIENT = (1 << 5)
AST_SSL_TLSV1_CLIENT = (1 << 5),
/*! Use server cipher order instead of the client order */
AST_SSL_SERVER_CIPHER_ORDER = (1 << 6),
/*! Disable TLSv1 support */
AST_SSL_DISABLE_TLSV1 = (1 << 7),
/*! Disable TLSv1.1 support */
AST_SSL_DISABLE_TLSV11 = (1 << 8),
/*! Disable TLSv1.2 support */
AST_SSL_DISABLE_TLSV12 = (1 << 9),
};
struct ast_tls_config {

View File

@@ -2102,10 +2102,13 @@ static int __ast_http_load(int reload)
}
http_tls_cfg.pvtfile = ast_strdup("");
/* Apply modern intermediate settings according to the Mozilla OpSec team as of July 30th, 2015 but disable TLSv1 */
ast_set_flag(&http_tls_cfg.flags, AST_SSL_DISABLE_TLSV1 | AST_SSL_SERVER_CIPHER_ORDER);
if (http_tls_cfg.cipher) {
ast_free(http_tls_cfg.cipher);
}
http_tls_cfg.cipher = ast_strdup("");
http_tls_cfg.cipher = ast_strdup("ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-DSS-AES128-GCM-SHA256:kEDH+AESGCM:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES128-SHA:ECDHE-ECDSA-AES128-SHA:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES256-SHA:DHE-RSA-AES128-SHA256:DHE-RSA-AES128-SHA:DHE-DSS-AES128-SHA256:DHE-RSA-AES256-SHA256:DHE-DSS-AES256-SHA:DHE-RSA-AES256-SHA:AES128-GCM-SHA256:AES256-GCM-SHA384:AES128-SHA256:AES256-SHA256:AES128-SHA:AES256-SHA:AES:CAMELLIA:DES-CBC3-SHA:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK:!aECDH:!EDH-DSS-DES-CBC3-SHA:!EDH-RSA-DES-CBC3-SHA:!KRB5-DES-CBC3-SHA");
AST_RWLIST_WRLOCK(&uri_redirects);
while ((redirect = AST_RWLIST_REMOVE_HEAD(&uri_redirects, entry))) {
@@ -2131,8 +2134,6 @@ static int __ast_http_load(int reload)
&& strcasecmp(v->name, "tlsdontverifyserver")
&& strcasecmp(v->name, "tlsclientmethod")
&& strcasecmp(v->name, "sslclientmethod")
&& strcasecmp(v->name, "tlscipher")
&& strcasecmp(v->name, "sslcipher")
&& !ast_tls_read_conf(&http_tls_cfg, &https_desc, v->name, v->value)) {
continue;
}

View File

@@ -759,7 +759,8 @@ static int __ssl_setup(struct ast_tls_config *cfg, int client)
return 0;
#else
int disable_ssl = 0;
long ssl_opts = 0;
if (!cfg->enabled) {
return 0;
}
@@ -807,12 +808,30 @@ static int __ssl_setup(struct ast_tls_config *cfg, int client)
* them. SSLv23_*_method supports TLSv1+.
*/
if (disable_ssl) {
long ssl_opts;
ssl_opts = SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
SSL_CTX_set_options(cfg->ssl_ctx, ssl_opts);
ssl_opts |= SSL_OP_NO_SSLv2 | SSL_OP_NO_SSLv3;
}
if (ast_test_flag(&cfg->flags, AST_SSL_SERVER_CIPHER_ORDER)) {
ssl_opts |= SSL_OP_CIPHER_SERVER_PREFERENCE;
}
if (ast_test_flag(&cfg->flags, AST_SSL_DISABLE_TLSV1)) {
ssl_opts |= SSL_OP_NO_TLSv1;
}
#if defined(HAVE_SSL_OP_NO_TLSV1_1) && defined(HAVE_SSL_OP_NO_TLSV1_2)
if (ast_test_flag(&cfg->flags, AST_SSL_DISABLE_TLSV11)) {
ssl_opts |= SSL_OP_NO_TLSv1_1;
}
if (ast_test_flag(&cfg->flags, AST_SSL_DISABLE_TLSV12)) {
ssl_opts |= SSL_OP_NO_TLSv1_2;
}
#else
ast_log(LOG_WARNING, "Your version of OpenSSL leaves you potentially vulnerable "
"to the SSL BEAST attack. Please upgrade to OpenSSL 1.0.1 or later\n");
#endif
SSL_CTX_set_options(cfg->ssl_ctx, ssl_opts);
SSL_CTX_set_verify(cfg->ssl_ctx,
ast_test_flag(&cfg->flags, AST_SSL_VERIFY_CLIENT) ? SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT : SSL_VERIFY_NONE,
NULL);
@@ -1164,6 +1183,14 @@ int ast_tls_read_conf(struct ast_tls_config *tls_cfg, struct ast_tcptls_session_
ast_clear_flag(&tls_cfg->flags, AST_SSL_TLSV1_CLIENT);
ast_clear_flag(&tls_cfg->flags, AST_SSL_SSLV3_CLIENT);
}
} else if (!strcasecmp(varname, "tlsservercipherorder")) {
ast_set2_flag(&tls_cfg->flags, ast_true(value), AST_SSL_SERVER_CIPHER_ORDER);
} else if (!strcasecmp(varname, "tlsdisablev1")) {
ast_set2_flag(&tls_cfg->flags, ast_true(value), AST_SSL_DISABLE_TLSV1);
} else if (!strcasecmp(varname, "tlsdisablev11")) {
ast_set2_flag(&tls_cfg->flags, ast_true(value), AST_SSL_DISABLE_TLSV11);
} else if (!strcasecmp(varname, "tlsdisablev12")) {
ast_set2_flag(&tls_cfg->flags, ast_true(value), AST_SSL_DISABLE_TLSV12);
} else {
return -1;
}

View File

@@ -305,16 +305,15 @@ static int decode_open_type(uint8_t *buf, unsigned int limit, unsigned int *len,
if (decode_length(buf, limit, len, &octet_cnt) != 0)
return -1;
if (octet_cnt > 0) {
/* Make sure the buffer contains at least the number of bits requested */
if ((*len + octet_cnt) > limit)
return -1;
*p_num_octets = octet_cnt;
*p_object = &buf[*len];
*len += octet_cnt;
/* Make sure the buffer contains at least the number of bits requested */
if ((*len + octet_cnt) > limit) {
return -1;
}
*p_num_octets = octet_cnt;
*p_object = &buf[*len];
*len += octet_cnt;
return 0;
}
/*- End of function --------------------------------------------------------*/

View File

@@ -223,7 +223,11 @@ static void sorcery_realtime_retrieve_regex(const struct ast_sorcery *sorcery, v
/* The realtime API provides no direct ability to do regex so for now we support a limited subset using pattern matching */
snprintf(field, sizeof(field), "%s LIKE", UUID_FIELD);
snprintf(value, sizeof(value), "%%%s%%", regex);
if (regex[0] == '^') {
snprintf(value, sizeof(value), "%s%%", regex + 1);
} else {
snprintf(value, sizeof(value), "%%%s%%", regex);
}
if (!(fields = ast_variable_new(field, value, ""))) {
return;