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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
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<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-13.1.0-rc1</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-13.1.0-rc1</h3>
<h3 align="center">Date: 2014-12-08</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="#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><br/><p>This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series 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-13.0.0.</p>
<hr/>
<a name="contributors"><h2 align="center">Contributors</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><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 closed by commits that went into this release.</p>
<table width="100%" border="0">
<tr>
<td width="33%"><h3>Coders</h3></td>
<td width="33%"><h3>Testers</h3></td>
<td width="33%"><h3>Reporters</h3></td>
</tr>
<tr valign="top">
<td>
23 mjordan<br/>
22 coreyfarrell<br/>
10 jcolp<br/>
9 rmudgett<br/>
8 gtjoseph<br/>
6 mmichelson<br/>
5 kharwell<br/>
4 file<br/>
3 jrose<br/>
3 mdavenport<br/>
2 kmoore<br/>
2 Nuno Borges<br/>
2 tzafrir<br/>
1 abelbeck<br/>
1 Birger Harzenetter<br/>
1 David Duncan Ross Palmer<br/>
1 Dmitriy Bubnov<br/>
1 Dmitry Bubnov<br/>
1 Etienne Lessard<br/>
1 igorg<br/>
1 jbigelow<br/>
1 oej<br/>
1 seanbright<br/>
1 sgriepentrog<br/>
1 wdoekes<br/>
1 Xavier Hienne<br/>
</td>
<td>
2 Beppo Maazucato<br/>
2 Gregory Malsack<br/>
1 David Duncan Ross Palmer<br/>
1 Etienne Lessard<br/>
1 ibercom<br/>
1 Nick Adams<br/>
1 xrobau<br/>
1 Zane Conkle<br/>
</td>
<td>
15 coreyfarrell<br/>
7 mjordan<br/>
2 beppo.it<br/>
2 hexanol<br/>
2 nerbos<br/>
2 rnewton<br/>
2 sgriepentrog<br/>
1 abelbeck<br/>
1 agupta<br/>
1 bensmithurst<br/>
1 dafi<br/>
1 davidw<br/>
1 dhanapathy<br/>
1 dmitriy.bubnov<br/>
1 gmalsack<br/>
1 gtj<br/>
1 jbigelow<br/>
1 jcolp<br/>
1 kharwell<br/>
1 laimbock<br/>
1 ldardini<br/>
1 m6kvm<br/>
1 mshepherd<br/>
1 Narkov<br/>
1 oej<br/>
1 rmudgett<br/>
1 snuffy<br/>
1 spitts<br/>
1 tzafrir<br/>
1 wimpy<br/>
1 xhienne<br/>
1 xrobau<br/>
1 yaronna<br/>
1 zconkle<br/>
1 zogot<br/>
</td>
</tr>
</table>
<hr/>
<a name="issues"><h2 align="center">Closed Issues</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
<h3>Category: Addons/chan_mobile</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24468">ASTERISK-24468</a>: Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427611">427611</a><br/>
Reporter: dmitriy.bubnov<br/>
Coders: Dmitriy Bubnov, Dmitry Bubnov<br/>
<br/>
<h3>Category: Applications/app_agent_pool</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24257">ASTERISK-24257</a>: agent must dial acceptdtmf twice to bridge to queue caller<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427512">427512</a><br/>
Reporter: spitts<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Applications/app_confbridge</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24522">ASTERISK-24522</a>: ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428079">428079</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Applications/app_dial</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428008">428008</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Applications/app_meetme</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24572">ASTERISK-24572</a>: [patch]App_meetme is loaded without its defaults when the configuration file is missing<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429029">429029</a><br/>
Reporter: nerbos<br/>
Coders: Nuno Borges<br/>
<br/>
<h3>Category: Applications/app_queue</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24454">ASTERISK-24454</a>: app_queue: ao2_iterator not destroyed, causing leak<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426266">426266</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24466">ASTERISK-24466</a>: app_queue: fix a couple leaks to struct call_queue<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426807">426807</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Applications/app_record</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24530">ASTERISK-24530</a>: [patch] app_record stripping 1/4 second from recordings<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428655">428655</a><br/>
Reporter: bensmithurst<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Applications/app_voicemail</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24190">ASTERISK-24190</a>: IMAP voicemail causes segfault<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426702">426702</a><br/>
Reporter: Narkov<br/>
Testers: Nick Adams<br/>
Coders: wdoekes<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24250">ASTERISK-24250</a>: [patch] Voicemail with multi-recipients To: header fix<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427585">427585</a><br/>
Reporter: abelbeck<br/>
Coders: abelbeck<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24476">ASTERISK-24476</a>: main/app.c / app_voicemail: ast_writestream leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427026">427026</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Applications/app_voicemail/IMAP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24190">ASTERISK-24190</a>: IMAP voicemail causes segfault<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426702">426702</a><br/>
Reporter: Narkov<br/>
Testers: Nick Adams<br/>
Coders: wdoekes<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24516">ASTERISK-24516</a>: [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428865">428865</a><br/>
Reporter: m6kvm<br/>
Testers: David Duncan Ross Palmer<br/>
Coders: David Duncan Ross Palmer<br/>
<br/>
<h3>Category: CDR/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24279">ASTERISK-24279</a>: Documentation: Clarify the behaviour of the CDR property 'unanswered'<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427902">427902</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<h3>Category: CEL/cel_odbc</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24283">ASTERISK-24283</a>: [patch]Microseconds precision in the eventtime column in the cel_odbc module<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427954">427954</a><br/>
Reporter: hexanol<br/>
Coders: Etienne Lessard<br/>
<br/>
<h3>Category: Channels/chan_mgcp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24500">ASTERISK-24500</a>: Regression introduced in chan_mgcp by SVN revision r227276<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427615">427615</a><br/>
Reporter: xhienne<br/>
Coders: Xavier Hienne<br/>
<br/>
<h3>Category: Channels/chan_phone</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24458">ASTERISK-24458</a>: chan_phone fails to build on big endian systems<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426570">426570</a><br/>
Reporter: tzafrir<br/>
Coders: tzafrir<br/>
<br/>
<h3>Category: Channels/chan_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24556">ASTERISK-24556</a>: Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension <br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429089">429089</a><br/>
Reporter: agupta<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24307">ASTERISK-24307</a>: Unintentional memory retention in stringfields<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427384">427384</a><br/>
Reporter: hexanol<br/>
Testers: ibercom, Etienne Lessard<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24533">ASTERISK-24533</a>: 2 threads created per chan_sip entry<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428687">428687</a><br/>
Reporter: xrobau<br/>
Testers: xrobau<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Channels/chan_sip/Interoperability</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21721">ASTERISK-21721</a>: SIP Failed to parse multiple Supported: headers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426597">426597</a><br/>
Reporter: oej<br/>
Coders: oej<br/>
<br/>
<h3>Category: Channels/chan_sip/Transfers</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15242">ASTERISK-15242</a>: transmit_refer leaks sip_refer structures<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428119">428119</a><br/>
Reporter: davidw<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Channels/chan_unistim</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24304">ASTERISK-24304</a>: asterisk crashing randomly because of unistim channel<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426668">426668</a><br/>
Reporter: dhanapathy<br/>
Coders: igorg<br/>
<br/>
<h3>Category: Contrib/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24432">ASTERISK-24432</a>: Install refcounter.py when REF_DEBUG is enabled<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426833">426833</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/AstMM</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24307">ASTERISK-24307</a>: Unintentional memory retention in stringfields<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427384">427384</a><br/>
Reporter: hexanol<br/>
Testers: ibercom, Etienne Lessard<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24535">ASTERISK-24535</a>: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428273">428273</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/Bridging</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24437">ASTERISK-24437</a>: Review implementation of ast_bridge_impart for leaks and document proper usage<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426432">426432</a><br/>
Reporter: sgriepentrog<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24447">ASTERISK-24447</a>: Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427494">427494</a><br/>
Reporter: rmudgett<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Core/BuildSystem</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24502">ASTERISK-24502</a>: Build fails when dev-mode, dont optimize and coverage are enabled<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427684">427684</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/Channels</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24542">ASTERISK-24542</a>: [patch]Failure showing codecs via 'core show channeltype <tech>'<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428632">428632</a><br/>
Reporter: snuffy<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20127">ASTERISK-20127</a>: [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427276">427276</a><br/>
Reporter: gtj<br/>
Coders: gtjoseph<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23651">ASTERISK-23651</a>: Reloading some modules that are loaded already, results in 'No such module' before a successful reload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427982">427982</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24487">ASTERISK-24487</a>: configuration: sections should be loadable as template even when not marked<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427228">427228</a><br/>
Reporter: sgriepentrog<br/>
Coders: gtjoseph<br/>
<br/>
<h3>Category: Core/FileFormatInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24492">ASTERISK-24492</a>: main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427466">427466</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23651">ASTERISK-23651</a>: Reloading some modules that are loaded already, results in 'No such module' before a successful reload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427982">427982</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24476">ASTERISK-24476</a>: main/app.c / app_voicemail: ast_writestream leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427026">427026</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/ManagerInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24430">ASTERISK-24430</a>: missing letter "p" in word response in OriginateResponse event documentation<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426368">426368</a><br/>
Reporter: dafi<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24453">ASTERISK-24453</a>: manager: acl_change_sub leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426525">426525</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24505">ASTERISK-24505</a>: manager: http connections leak references<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427643">427643</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24554">ASTERISK-24554</a>: AMI/ARI: Generate events on connected line changes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429064">429064</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Core/Netsock</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24469">ASTERISK-24469</a>: Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428425">428425</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Core/PBX</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24444">ASTERISK-24444</a>: PBX: Crash when generating extension for pattern matching hint<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427711">427711</a><br/>
Reporter: ldardini<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/RTP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24489">ASTERISK-24489</a>: Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427763">427763</a><br/>
Reporter: gmalsack<br/>
Testers: Gregory Malsack, Beppo Maazucato<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Documentation</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24430">ASTERISK-24430</a>: missing letter "p" in word response in OriginateResponse event documentation<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426368">426368</a><br/>
Reporter: dafi<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Functions/func_cdr</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24455">ASTERISK-24455</a>: func_cdr: CDR_PROP leaks payload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426252">426252</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Functions/func_talkdetect</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24482">ASTERISK-24482</a>: func_talkdetect: Fix stasis message leak in audiohook callback<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427204">427204</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: PBX/pbx_loopback</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24577">ASTERISK-24577</a>: Speed up loopback switches by avoiding unneeded lookups<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428789">428789</a><br/>
Reporter: wimpy<br/>
Coders: Birger Harzenetter<br/>
<br/>
<h3>Category: Resources/res_ari</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24501">ASTERISK-24501</a>: ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427789">427789</a><br/>
Reporter: mjordan<br/>
Coders: kmoore<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24554">ASTERISK-24554</a>: AMI/ARI: Generate events on connected line changes<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429064">429064</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Resources/res_fax</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24457">ASTERISK-24457</a>: res_fax: fax gateway frames leak<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426529">426529</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Resources/res_hep</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24491">ASTERISK-24491</a>: Memory leak in res_hep<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427405">427405</a><br/>
Reporter: zconkle<br/>
Testers: Zane Conkle<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Resources/res_hep_rtcp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24489">ASTERISK-24489</a>: Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427763">427763</a><br/>
Reporter: gmalsack<br/>
Testers: Gregory Malsack, Beppo Maazucato<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24498">ASTERISK-24498</a>: Segmentation fault in res_hep_rtcp on attended transfer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427763">427763</a><br/>
Reporter: beppo.it<br/>
Testers: Gregory Malsack, Beppo Maazucato<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_http_websocket</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24480">ASTERISK-24480</a>: res_http_websockets: Module reference decrease below zero<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427201">427201</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Resources/res_monitor</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24573">ASTERISK-24573</a>: [patch]Out of sync conversation recording when divided in multiple recordings<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429033">429033</a><br/>
Reporter: nerbos<br/>
Coders: Nuno Borges<br/>
<br/>
<h3>Category: Resources/res_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24336">ASTERISK-24336</a>: PJSIP timer_min_se value under 90 causes crash<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427979">427979</a><br/>
Reporter: zogot<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24462">ASTERISK-24462</a>: res_pjsip: Stale qualify statistics after disablementation<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426757">426757</a><br/>
Reporter: kharwell<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24471">ASTERISK-24471</a>: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428302">428302</a><br/>
Reporter: yaronna<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428008">428008</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24508">ASTERISK-24508</a>: pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428196">428196</a><br/>
Reporter: beppo.it<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_pjsip_acl</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24531">ASTERISK-24531</a>: res_pjsip_acl: ACLs not applied on initial module load<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428343">428343</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Resources/res_pjsip_multihomed</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24438">ASTERISK-24438</a>: res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427303">427303</a><br/>
Reporter: mshepherd<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_pjsip_outbound_registration</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24411">ASTERISK-24411</a>: [patch] Status of outbound registration is not changed upon unregistering.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426924">426924</a><br/>
Reporter: jbigelow<br/>
Coders: jbigelow<br/>
<br/>
<h3>Category: Resources/res_pjsip_refer</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24508">ASTERISK-24508</a>: pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428196">428196</a><br/>
Reporter: beppo.it<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24528">ASTERISK-24528</a>: res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428305">428305</a><br/>
Reporter: jcolp<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_srtp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24436">ASTERISK-24436</a>: Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426143">426143</a><br/>
Reporter: laimbock<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_stasis</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24537">ASTERISK-24537</a>: Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429062">429062</a><br/>
Reporter: mjordan<br/>
Coders: kmoore<br/>
<br/>
<h3>Category: pjproject/pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24336">ASTERISK-24336</a>: PJSIP timer_min_se value under 90 causes crash<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427979">427979</a><br/>
Reporter: zogot<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24471">ASTERISK-24471</a>: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428302">428302</a><br/>
Reporter: yaronna<br/>
Coders: jcolp<br/>
<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><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
<table width="100%" border="1">
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426120">426120</a></td><td>jrose</td><td>Documentation: Improve documentation for ExtensionStatus AMI events</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426176">426176</a></td><td>mjordan</td><td>res/res_phoneprov: Fix crash on shutdown caused by container cleanup</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426211">426211</a></td><td>mjordan</td><td>res/res_http_websocket: Fix minor nits found by wdoekes on r409681</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426234">426234</a></td><td>seanbright</td><td>configure: Add autoconf check for libopus.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426294">426294</a></td><td>mdavenport</td><td>ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426362">426362</a></td><td>mdavenport</td><td>ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426459">426459</a></td><td>mdavenport</td><td>ASTERISK-23512, correct inaccurate comment in manager.conf.sample</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426552">426552</a></td><td>rmudgett</td><td>bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config().</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426602">426602</a></td><td>mjordan</td><td>channels/chan_sip: Add improved support for 4xx error codes</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426780">426780</a></td><td>kharwell</td><td>res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426865">426865</a></td><td>mjordan</td><td>channels/sip/reqresp_parser: Fix unit tests for r426594</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426934">426934</a></td><td>tzafrir</td><td>install init.d files on GNU/kFreeBSD</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=426996">426996</a></td><td>mjordan</td><td>res/res_stasis: Fix crash on module unload while performing operation</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427021">427021</a></td><td>coreyfarrell</td><td>func_jitterbuffer: fix frame leaks.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427089">427089</a></td><td>coreyfarrell</td><td>Fix compile error caused by review 4138</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427130">427130</a></td><td>rmudgett</td><td>res_pjsip: Add disable_tcp_switch option.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427181">427181</a></td><td>coreyfarrell</td><td>Fix crash caused by merge error on review 4138</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427334">427334</a></td><td>mmichelson</td><td>Make the disable_tcp_switch PJSIP system object enabled by default.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427356">427356</a></td><td>gtjoseph</td><td>test_strings: Remove string tests that exercise asserts.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427583">427583</a></td><td>mjordan</td><td>bridge_native_rtp: Fix T.38 issues with remote bridges</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427737">427737</a></td><td>coreyfarrell</td><td>Fix leak in AMI Action Bridge</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427815">427815</a></td><td>kharwell</td><td>res_pjsip_exten_state: PJSIPShowSubscriptionsInbound causes crash</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427841">427841</a></td><td>mmichelson</td><td>Fix race condition where duplicated requests may be handled by multiple threads.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427846">427846</a></td><td>file</td><td>app_confbridge: Play "leader has left" sound even when musiconhold is enabled.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427870">427870</a></td><td>mmichelson</td><td>Fix race condition that could result in ARI transfer messages not being sent.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427876">427876</a></td><td>sgriepentrog</td><td>stun: correct attribute string padding to match rfc</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=427927">427927</a></td><td>mjordan</td><td>tests/test_cel: Unlock bridge on off nominal paths</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428052">428052</a></td><td>file</td><td>chan_pjsip: Remove AOR check when dialing and one is specified.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428115">428115</a></td><td>mjordan</td><td>apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428145">428145</a></td><td>mmichelson</td><td>Allow for transferer to retry when dialing an invalid extension.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428169">428169</a></td><td>rmudgett</td><td>parking_tests.c: Add missing newline on a unit test message.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428222">428222</a></td><td>file</td><td>res_pjsip_sdp_rtp: Add support for optimistic SRTP.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428246">428246</a></td><td>rmudgett</td><td>ast_str: Fix improper member access to struct ast_str members.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428339">428339</a></td><td>kharwell</td><td>AST-2014-017 - app_confbridge: permission escalation/ class authorization.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428413">428413</a></td><td>kharwell</td><td>AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428505">428505</a></td><td>mjordan</td><td>main/bridge_basic: Fix features regressions introduced by r428165</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428544">428544</a></td><td>gtjoseph</td><td>sorcery: Make is_object_field_registered handle field names that are regexes.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428572">428572</a></td><td>rmudgett</td><td>manager: Fix could not extend string messages.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428602">428602</a></td><td>rmudgett</td><td>DTMF hooks: Leaving channels need to push any collected digits into the bridge.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428604">428604</a></td><td>rmudgett</td><td>test_channel_feature_hooks.c: Fix unit test for DTMF hooks.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428731">428731</a></td><td>gtjoseph</td><td>res_pjsip_endpoint_identifier_ip: Add 'show identify(ies)' cli commands</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428734">428734</a></td><td>gtjoseph</td><td>config: Create ast_variable_find_in_list()</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428761">428761</a></td><td>file</td><td>res_pjsip_refer: Fix issue where native bridge may not occur upon completion of a transfer.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428815">428815</a></td><td>mjordan</td><td>tests/test_stasis: Resolve compilation issues from Asterisk 12 merge</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428837">428837</a></td><td>gtjoseph</td><td>CHANGES: Add item for new 'pjsip show identif(y|ies) commands</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428892">428892</a></td><td>mjordan</td><td>tests/test_cel: Fix CEL unit test failures caused by attended transfer changes</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428919">428919</a></td><td>mjordan</td><td>tests/test_cel: Add test_cel_attended_transfer_bridges_link to racey tests</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428946">428946</a></td><td>mjordan</td><td>main/test: Fix race condition between AMI topic and Test Suite topic</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=428973">428973</a></td><td>mjordan</td><td>main/test: Fix compilation issue on 32-bit systems</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429000">429000</a></td><td>gtjoseph</td><td>sorcery: Add additional observer capabilities.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=429091">429091</a></td><td>mjordan</td><td>AMI/ARI: Update version to 2.6.0/1.6.0 respectively for new features</td>
<td></td></tr></table>
<hr/>
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
<center><a href="#top">[Back to Top]</a></center><br/><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>
CHANGES | 43
Makefile | 4
Makefile.rules | 18
UPGRADE.txt | 25
addons/chan_mobile.c | 2
apps/app_agent_pool.c | 39
apps/app_confbridge.c | 4
apps/app_meetme.c | 35
apps/app_queue.c | 15
apps/app_record.c | 9
apps/app_voicemail.c | 45
apps/confbridge/conf_state_multi_marked.c | 27
bridges/bridge_builtin_features.c | 4
bridges/bridge_native_rtp.c | 10
build_tools/menuselect-deps.in | 1
cel/cel_odbc.c | 13
channels/chan_console.c | 64 -
channels/chan_dahdi.c | 2
channels/chan_iax2.c | 2
channels/chan_mgcp.c | 7
channels/chan_motif.c | 6
channels/chan_phone.c | 2
channels/chan_pjsip.c | 20
channels/chan_sip.c | 113 +-
channels/chan_skinny.c | 2
channels/chan_unistim.c | 8
channels/sig_pri.c | 2
channels/sip/include/reqresp_parser.h | 5
channels/sip/reqresp_parser.c | 6
channels/sip/security_events.c | 2
configs/samples/cdr.conf.sample | 21
configs/samples/extensions.conf.sample | 2
configs/samples/features.conf.sample | 4
configs/samples/manager.conf.sample | 2
configs/samples/pjsip.conf.sample | 13
configs/samples/stasis.conf.sample | 10
configure.ac | 3
contrib/Makefile | 29
contrib/ast-db-manage/config/versions/eb88a14f2a_add_media_encryption_optimistic_to_pjsip.py | 31
funcs/func_cdr.c | 3
funcs/func_db.c | 2
funcs/func_talkdetect.c | 1
include/asterisk/autoconfig.h.in | 3
include/asterisk/bridge_channel.h | 25
include/asterisk/channel.h | 11
include/asterisk/config.h | 14
include/asterisk/features_config.h | 6
include/asterisk/manager.h | 2
include/asterisk/res_pjsip.h | 5
include/asterisk/res_pjsip_session.h | 2
include/asterisk/sorcery.h | 176 +++
include/asterisk/stasis.h | 36
include/asterisk/stasis_app.h | 5
include/asterisk/stasis_bridges.h | 213 ++--
include/asterisk/stasis_channels.h | 14
include/asterisk/stasis_internal.h | 7
include/asterisk/stasis_message_router.h | 16
include/asterisk/stringfields.h | 47 -
include/asterisk/test.h | 4
include/asterisk/utils.h | 10
main/abstract_jb.c | 15
main/acl.c | 2
main/app.c | 18
main/audiohook.c | 2
main/bridge.c | 283 +-----
main/bridge_basic.c | 185 +---
main/bridge_channel.c | 359 +++++--
main/cdr.c | 22
main/cel.c | 4
main/channel.c | 6
main/channel_internal_api.c | 9
main/config.c | 32
main/endpoints.c | 2
main/features.c | 1
main/features_config.c | 34
main/file.c | 4
main/manager.c | 185 ++--
main/manager_channels.c | 20
main/pbx.c | 26
main/rtp_engine.c | 2
main/sorcery.c | 365 ++++++-
main/stasis.c | 152 ++-
main/stasis_bridges.c | 264 ++---
main/stasis_cache.c | 2
main/stasis_channels.c | 10
main/stasis_message_router.c | 22
main/stun.c | 11
main/test.c | 50 -
main/utils.c | 46
makeopts.in | 3
pbx/pbx_config.c | 31
pbx/pbx_loopback.c | 19
res/ari/ari_model_validators.c | 85 +
res/ari/ari_model_validators.h | 23
res/parking/parking_applications.c | 2
res/parking/parking_bridge_features.c | 2
res/parking/parking_tests.c | 2
res/res_agi.c | 4
res/res_calendar.c | 2
res/res_fax.c | 25
res/res_hep.c | 1
res/res_http_websocket.c | 18
res/res_monitor.c | 2
res/res_phoneprov.c | 6
res/res_pjsip.c | 35
res/res_pjsip/config_system.c | 8
res/res_pjsip/pjsip_cli.c | 5
res/res_pjsip/pjsip_configuration.c | 9
res/res_pjsip/pjsip_distributor.c | 28
res/res_pjsip/pjsip_options.c | 15
res/res_pjsip_acl.c | 7
res/res_pjsip_endpoint_identifier_ip.c | 83 +
res/res_pjsip_exten_state.c | 4
res/res_pjsip_multihomed.c | 8
res/res_pjsip_mwi.c | 2
res/res_pjsip_outbound_registration.c | 461 ++++++----
res/res_pjsip_phoneprov_provider.c | 9
res/res_pjsip_pubsub.c | 24
res/res_pjsip_refer.c | 63 +
res/res_pjsip_sdp_rtp.c | 105 +-
res/res_pjsip_session.c | 42
res/res_srtp.c | 1
res/res_stasis.c | 314 +++---
res/res_stasis_device_state.c | 2
res/res_xmpp.c | 2
res/stasis/app.c | 59 -
res/stasis/app.h | 7
res/stasis/stasis_bridge.c | 6
rest-api/api-docs/applications.json | 2
rest-api/api-docs/asterisk.json | 2
rest-api/api-docs/bridges.json | 2
rest-api/api-docs/channels.json | 2
rest-api/api-docs/deviceStates.json | 2
rest-api/api-docs/endpoints.json | 2
rest-api/api-docs/events.json | 16
rest-api/api-docs/mailboxes.json | 2
rest-api/api-docs/playbacks.json | 2
rest-api/api-docs/recordings.json | 2
rest-api/api-docs/sounds.json | 2
rest-api/resources.json | 2
tests/test_cel.c | 72 -
tests/test_channel_feature_hooks.c | 31
tests/test_sorcery.c | 355 +++++++
tests/test_stasis.c | 310 ++++++
tests/test_strings.c | 66 +
145 files changed, 4106 insertions(+), 1675 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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asterisk-13.1.0-rc1-summary.txt
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Release Summary
asterisk-13.1.0-rc1
Date: 2014-12-08
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release includes only bug fixes. The changes included were made only
to address problems that have been identified in this release series.
Users should be able to safely upgrade to this version if this release
series is already in use. Users considering upgrading from a previous
release series 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-13.0.0.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were closed by commits that went into this
release.
Coders Testers Reporters
23 mjordan 2 Beppo Maazucato 15 coreyfarrell
22 coreyfarrell 2 Gregory Malsack 7 mjordan
10 jcolp 1 David Duncan Ross Palmer 2 beppo.it
9 rmudgett 1 Etienne Lessard 2 hexanol
8 gtjoseph 1 ibercom 2 nerbos
6 mmichelson 1 Nick Adams 2 rnewton
5 kharwell 1 xrobau 2 sgriepentrog
4 file 1 Zane Conkle 1 abelbeck
3 jrose 1 agupta
3 mdavenport 1 bensmithurst
2 kmoore 1 dafi
2 Nuno Borges 1 davidw
2 tzafrir 1 dhanapathy
1 abelbeck 1 dmitriy.bubnov
1 Birger Harzenetter 1 gmalsack
1 David Duncan Ross Palmer 1 gtj
1 Dmitriy Bubnov 1 jbigelow
1 Dmitry Bubnov 1 jcolp
1 Etienne Lessard 1 kharwell
1 igorg 1 laimbock
1 jbigelow 1 ldardini
1 oej 1 m6kvm
1 seanbright 1 mshepherd
1 sgriepentrog 1 Narkov
1 wdoekes 1 oej
1 Xavier Hienne 1 rmudgett
1 snuffy
1 spitts
1 tzafrir
1 wimpy
1 xhienne
1 xrobau
1 yaronna
1 zconkle
1 zogot
----------------------------------------------------------------------
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.
Category: Addons/chan_mobile
ASTERISK-24468: Incoming UCS2 encoded SMS truncated if SMS length exceeds
50 (roughly) national symbols
Revision: 427611
Reporter: dmitriy.bubnov
Coders: Dmitriy Bubnov, Dmitry Bubnov
Category: Applications/app_agent_pool
ASTERISK-24257: agent must dial acceptdtmf twice to bridge to queue caller
Revision: 427512
Reporter: spitts
Coders: rmudgett
Category: Applications/app_confbridge
ASTERISK-24522: ConfBridge: delay occurs between kicking all endmarked
users when last marked user leaves
Revision: 428079
Reporter: mjordan
Coders: mjordan
Category: Applications/app_dial
ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Revision: 428008
Reporter: rnewton
Coders: jcolp
Category: Applications/app_meetme
ASTERISK-24572: [patch]App_meetme is loaded without its defaults when the
configuration file is missing
Revision: 429029
Reporter: nerbos
Coders: Nuno Borges
Category: Applications/app_queue
ASTERISK-24454: app_queue: ao2_iterator not destroyed, causing leak
Revision: 426266
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24466: app_queue: fix a couple leaks to struct call_queue
Revision: 426807
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Applications/app_record
ASTERISK-24530: [patch] app_record stripping 1/4 second from recordings
Revision: 428655
Reporter: bensmithurst
Coders: jcolp
Category: Applications/app_voicemail
ASTERISK-24190: IMAP voicemail causes segfault
Revision: 426702
Reporter: Narkov
Testers: Nick Adams
Coders: wdoekes
ASTERISK-24250: [patch] Voicemail with multi-recipients To: header fix
Revision: 427585
Reporter: abelbeck
Coders: abelbeck
ASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks
Revision: 427026
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Applications/app_voicemail/IMAP
ASTERISK-24190: IMAP voicemail causes segfault
Revision: 426702
Reporter: Narkov
Testers: Nick Adams
Coders: wdoekes
ASTERISK-24516: [patch]Asterisk segfaults when playing back voicemail
under high concurrency with an IMAP backend
Revision: 428865
Reporter: m6kvm
Testers: David Duncan Ross Palmer
Coders: David Duncan Ross Palmer
Category: CDR/General
ASTERISK-24279: Documentation: Clarify the behaviour of the CDR property
'unanswered'
Revision: 427902
Reporter: mjordan
Coders: jrose
Category: CEL/cel_odbc
ASTERISK-24283: [patch]Microseconds precision in the eventtime column in
the cel_odbc module
Revision: 427954
Reporter: hexanol
Coders: Etienne Lessard
Category: Channels/chan_mgcp
ASTERISK-24500: Regression introduced in chan_mgcp by SVN revision r227276
Revision: 427615
Reporter: xhienne
Coders: Xavier Hienne
Category: Channels/chan_phone
ASTERISK-24458: chan_phone fails to build on big endian systems
Revision: 426570
Reporter: tzafrir
Coders: tzafrir
Category: Channels/chan_pjsip
ASTERISK-24556: Asterisk 13 core dumps when calling from pjsip extension
to another pjsip extension
Revision: 429089
Reporter: agupta
Coders: mmichelson
Category: Channels/chan_sip/General
ASTERISK-24307: Unintentional memory retention in stringfields
Revision: 427384
Reporter: hexanol
Testers: ibercom, Etienne Lessard
Coders: coreyfarrell
ASTERISK-24533: 2 threads created per chan_sip entry
Revision: 428687
Reporter: xrobau
Testers: xrobau
Coders: mjordan
Category: Channels/chan_sip/Interoperability
ASTERISK-21721: SIP Failed to parse multiple Supported: headers
Revision: 426597
Reporter: oej
Coders: oej
Category: Channels/chan_sip/Transfers
ASTERISK-15242: transmit_refer leaks sip_refer structures
Revision: 428119
Reporter: davidw
Coders: coreyfarrell
Category: Channels/chan_unistim
ASTERISK-24304: asterisk crashing randomly because of unistim channel
Revision: 426668
Reporter: dhanapathy
Coders: igorg
Category: Contrib/General
ASTERISK-24432: Install refcounter.py when REF_DEBUG is enabled
Revision: 426833
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/AstMM
ASTERISK-24307: Unintentional memory retention in stringfields
Revision: 427384
Reporter: hexanol
Testers: ibercom, Etienne Lessard
Coders: coreyfarrell
ASTERISK-24535: stringfields: Fix regression from fix for unintentional
memory retention and another issue exposed by the fix
Revision: 428273
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/Bridging
ASTERISK-24437: Review implementation of ast_bridge_impart for leaks and
document proper usage
Revision: 426432
Reporter: sgriepentrog
Coders: mjordan
ASTERISK-24447: Bridge DTMF hooks: Audio doesn't pass when waiting for
more matching digits.
Revision: 427494
Reporter: rmudgett
Coders: rmudgett
Category: Core/BuildSystem
ASTERISK-24502: Build fails when dev-mode, dont optimize and coverage are
enabled
Revision: 427684
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/Channels
ASTERISK-24542: [patch]Failure showing codecs via 'core show channeltype '
Revision: 428632
Reporter: snuffy
Coders: jcolp
Category: Core/Configuration
ASTERISK-20127: [Regression] Config.c config_text_file_load() unescapes
semicolons ("\;" -> ";") turning them into comments (corruption) on
rewrite of a config file
Revision: 427276
Reporter: gtj
Coders: gtjoseph
ASTERISK-23651: Reloading some modules that are loaded already, results in
'No such module' before a successful reload
Revision: 427982
Reporter: rnewton
Coders: jcolp
ASTERISK-24487: configuration: sections should be loadable as template
even when not marked
Revision: 427228
Reporter: sgriepentrog
Coders: gtjoseph
Category: Core/FileFormatInterface
ASTERISK-24492: main/file.c: ast_filestream sometimes causes extra calls
to ast_module_unref
Revision: 427466
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/General
ASTERISK-23651: Reloading some modules that are loaded already, results in
'No such module' before a successful reload
Revision: 427982
Reporter: rnewton
Coders: jcolp
ASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks
Revision: 427026
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/ManagerInterface
ASTERISK-24430: missing letter "p" in word response in OriginateResponse
event documentation
Revision: 426368
Reporter: dafi
Coders: mjordan
ASTERISK-24453: manager: acl_change_sub leaks
Revision: 426525
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24505: manager: http connections leak references
Revision: 427643
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24554: AMI/ARI: Generate events on connected line changes
Revision: 429064
Reporter: mjordan
Coders: mmichelson
Category: Core/Netsock
ASTERISK-24469: Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked
addresses through
Revision: 428425
Reporter: mjordan
Coders: mjordan
Category: Core/PBX
ASTERISK-24444: PBX: Crash when generating extension for pattern matching
hint
Revision: 427711
Reporter: ldardini
Coders: jcolp
Category: Core/RTP
ASTERISK-24489: Crash: Asterisk crashes when converting RTCP packet to
JSON for res_hep_rtcp and report blocks are greater than 1
Revision: 427763
Reporter: gmalsack
Testers: Gregory Malsack, Beppo Maazucato
Coders: mjordan
Category: Documentation
ASTERISK-24430: missing letter "p" in word response in OriginateResponse
event documentation
Revision: 426368
Reporter: dafi
Coders: mjordan
Category: Functions/func_cdr
ASTERISK-24455: func_cdr: CDR_PROP leaks payload
Revision: 426252
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Functions/func_talkdetect
ASTERISK-24482: func_talkdetect: Fix stasis message leak in audiohook
callback
Revision: 427204
Reporter: coreyfarrell
Coders: coreyfarrell
Category: PBX/pbx_loopback
ASTERISK-24577: Speed up loopback switches by avoiding unneeded lookups
Revision: 428789
Reporter: wimpy
Coders: Birger Harzenetter
Category: Resources/res_ari
ASTERISK-24501: ARI: Moving a channel between bridges followed by a hangup
can cause an ARI client to not receive an expected ChannelLeftBridge event
before StasisEnd
Revision: 427789
Reporter: mjordan
Coders: kmoore
ASTERISK-24554: AMI/ARI: Generate events on connected line changes
Revision: 429064
Reporter: mjordan
Coders: mmichelson
Category: Resources/res_fax
ASTERISK-24457: res_fax: fax gateway frames leak
Revision: 426529
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Resources/res_hep
ASTERISK-24491: Memory leak in res_hep
Revision: 427405
Reporter: zconkle
Testers: Zane Conkle
Coders: coreyfarrell
Category: Resources/res_hep_rtcp
ASTERISK-24489: Crash: Asterisk crashes when converting RTCP packet to
JSON for res_hep_rtcp and report blocks are greater than 1
Revision: 427763
Reporter: gmalsack
Testers: Gregory Malsack, Beppo Maazucato
Coders: mjordan
ASTERISK-24498: Segmentation fault in res_hep_rtcp on attended transfer
Revision: 427763
Reporter: beppo.it
Testers: Gregory Malsack, Beppo Maazucato
Coders: mjordan
Category: Resources/res_http_websocket
ASTERISK-24480: res_http_websockets: Module reference decrease below zero
Revision: 427201
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Resources/res_monitor
ASTERISK-24573: [patch]Out of sync conversation recording when divided in
multiple recordings
Revision: 429033
Reporter: nerbos
Coders: Nuno Borges
Category: Resources/res_pjsip
ASTERISK-24336: PJSIP timer_min_se value under 90 causes crash
Revision: 427979
Reporter: zogot
Coders: jcolp
ASTERISK-24462: res_pjsip: Stale qualify statistics after disablementation
Revision: 426757
Reporter: kharwell
Coders: kharwell
ASTERISK-24471: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate
from /usr/local/lib/libpjmedia.so.2
Revision: 428302
Reporter: yaronna
Coders: jcolp
ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Revision: 428008
Reporter: rnewton
Coders: jcolp
ASTERISK-24508: pjsip - REFER request from SNOM is rejected with "400 bad
request" - DEBUG shows "Received a REFER without a parseable Refer-To"
Revision: 428196
Reporter: beppo.it
Coders: jcolp
Category: Resources/res_pjsip_acl
ASTERISK-24531: res_pjsip_acl: ACLs not applied on initial module load
Revision: 428343
Reporter: mjordan
Coders: jrose
Category: Resources/res_pjsip_multihomed
ASTERISK-24438: res_pjsip_multihomed.so blocks Asterisk reload when DNS
settings invalid
Revision: 427303
Reporter: mshepherd
Coders: jcolp
Category: Resources/res_pjsip_outbound_registration
ASTERISK-24411: [patch] Status of outbound registration is not changed
upon unregistering.
Revision: 426924
Reporter: jbigelow
Coders: jbigelow
Category: Resources/res_pjsip_refer
ASTERISK-24508: pjsip - REFER request from SNOM is rejected with "400 bad
request" - DEBUG shows "Received a REFER without a parseable Refer-To"
Revision: 428196
Reporter: beppo.it
Coders: jcolp
ASTERISK-24528: res_pjsip_refer: Sending INVITE with Replaces in-dialog
with invalid target causes crash
Revision: 428305
Reporter: jcolp
Coders: jcolp
Category: Resources/res_srtp
ASTERISK-24436: Missing header in res/res_srtp.c when compiling against
libsrtp-1.5.0
Revision: 426143
Reporter: laimbock
Coders: mjordan
Category: Resources/res_stasis
ASTERISK-24537: Stasis: StasisStart/StasisEnd events are not reliably
transmitted during transfers
Revision: 429062
Reporter: mjordan
Coders: kmoore
Category: pjproject/pjsip
ASTERISK-24336: PJSIP timer_min_se value under 90 causes crash
Revision: 427979
Reporter: zogot
Coders: jcolp
ASTERISK-24471: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate
from /usr/local/lib/libpjmedia.so.2
Revision: 428302
Reporter: yaronna
Coders: jcolp
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
directly close an issue from the issue tracker. The commits may have been
marked as being related to an issue. If that is the case, the issue
numbers are listed here, as well.
+------------------------------------------------------------------------+
|Revision|Author |Summary |Issues |
| | | |Referenced|
|--------+------------+---------------------------------------+----------|
|426120 |jrose |Documentation: Improve documentation | |
| | |for ExtensionStatus AMI events | |
|--------+------------+---------------------------------------+----------|
|426176 |mjordan |res/res_phoneprov: Fix crash on | |
| | |shutdown caused by container cleanup | |
|--------+------------+---------------------------------------+----------|
|426211 |mjordan |res/res_http_websocket: Fix minor nits | |
| | |found by wdoekes on r409681 | |
|--------+------------+---------------------------------------+----------|
|426234 |seanbright |configure: Add autoconf check for | |
| | |libopus. | |
|--------+------------+---------------------------------------+----------|
| | |ASTERISK-24419, fix incorrect syntax | |
|426294 |mdavenport |for setting language in | |
| | |extensions.conf.sample | |
|--------+------------+---------------------------------------+----------|
| | |ASTERISK-24323, fix bug in | |
|426362 |mdavenport |documentation of AGI STREAM FILE | |
| | |CONTROL | |
|--------+------------+---------------------------------------+----------|
|426459 |mdavenport |ASTERISK-23512, correct inaccurate | |
| | |comment in manager.conf.sample | |
|--------+------------+---------------------------------------+----------|
| | |bridge_builtin_features: Add missing | |
|426552 |rmudgett |channel locks around | |
| | |ast_get_chan_features_general_config().| |
|--------+------------+---------------------------------------+----------|
|426602 |mjordan |channels/chan_sip: Add improved support| |
| | |for 4xx error codes | |
|--------+------------+---------------------------------------+----------|
| | |res_pjsip_exten_state: | |
|426780 |kharwell |PJSIPShowSubscriptionsInbound causes | |
| | |crash | |
|--------+------------+---------------------------------------+----------|
|426865 |mjordan |channels/sip/reqresp_parser: Fix unit | |
| | |tests for r426594 | |
|--------+------------+---------------------------------------+----------|
|426934 |tzafrir |install init.d files on GNU/kFreeBSD | |
|--------+------------+---------------------------------------+----------|
|426996 |mjordan |res/res_stasis: Fix crash on module | |
| | |unload while performing operation | |
|--------+------------+---------------------------------------+----------|
|427021 |coreyfarrell|func_jitterbuffer: fix frame leaks. | |
|--------+------------+---------------------------------------+----------|
|427089 |coreyfarrell|Fix compile error caused by review 4138| |
|--------+------------+---------------------------------------+----------|
|427130 |rmudgett |res_pjsip: Add disable_tcp_switch | |
| | |option. | |
|--------+------------+---------------------------------------+----------|
|427181 |coreyfarrell|Fix crash caused by merge error on | |
| | |review 4138 | |
|--------+------------+---------------------------------------+----------|
|427334 |mmichelson |Make the disable_tcp_switch PJSIP | |
| | |system object enabled by default. | |
|--------+------------+---------------------------------------+----------|
|427356 |gtjoseph |test_strings: Remove string tests that | |
| | |exercise asserts. | |
|--------+------------+---------------------------------------+----------|
|427583 |mjordan |bridge_native_rtp: Fix T.38 issues with| |
| | |remote bridges | |
|--------+------------+---------------------------------------+----------|
|427737 |coreyfarrell|Fix leak in AMI Action Bridge | |
|--------+------------+---------------------------------------+----------|
| | |res_pjsip_exten_state: | |
|427815 |kharwell |PJSIPShowSubscriptionsInbound causes | |
| | |crash | |
|--------+------------+---------------------------------------+----------|
| | |Fix race condition where duplicated | |
|427841 |mmichelson |requests may be handled by multiple | |
| | |threads. | |
|--------+------------+---------------------------------------+----------|
|427846 |file |app_confbridge: Play "leader has left" | |
| | |sound even when musiconhold is enabled.| |
|--------+------------+---------------------------------------+----------|
|427870 |mmichelson |Fix race condition that could result in| |
| | |ARI transfer messages not being sent. | |
|--------+------------+---------------------------------------+----------|
|427876 |sgriepentrog|stun: correct attribute string padding | |
| | |to match rfc | |
|--------+------------+---------------------------------------+----------|
|427927 |mjordan |tests/test_cel: Unlock bridge on off | |
| | |nominal paths | |
|--------+------------+---------------------------------------+----------|
|428052 |file |chan_pjsip: Remove AOR check when | |
| | |dialing and one is specified. | |
|--------+------------+---------------------------------------+----------|
| | |apps/app_confbridge: Ensure 'normal' | |
|428115 |mjordan |users hear message when last marked | |
| | |leaves | |
|--------+------------+---------------------------------------+----------|
|428145 |mmichelson |Allow for transferer to retry when | |
| | |dialing an invalid extension. | |
|--------+------------+---------------------------------------+----------|
|428169 |rmudgett |parking_tests.c: Add missing newline on| |
| | |a unit test message. | |
|--------+------------+---------------------------------------+----------|
|428222 |file |res_pjsip_sdp_rtp: Add support for | |
| | |optimistic SRTP. | |
|--------+------------+---------------------------------------+----------|
|428246 |rmudgett |ast_str: Fix improper member access to | |
| | |struct ast_str members. | |
|--------+------------+---------------------------------------+----------|
| | |AST-2014-017 - app_confbridge: | |
|428339 |kharwell |permission escalation/ class | |
| | |authorization. | |
|--------+------------+---------------------------------------+----------|
|428413 |kharwell |AST-2014-018 - func_db: DB Dialplan | |
| | |function permission escalation via AMI.| |
|--------+------------+---------------------------------------+----------|
|428505 |mjordan |main/bridge_basic: Fix features | |
| | |regressions introduced by r428165 | |
|--------+------------+---------------------------------------+----------|
| | |sorcery: Make | |
|428544 |gtjoseph |is_object_field_registered handle field| |
| | |names that are regexes. | |
|--------+------------+---------------------------------------+----------|
|428572 |rmudgett |manager: Fix could not extend string | |
| | |messages. | |
|--------+------------+---------------------------------------+----------|
| | |DTMF hooks: Leaving channels need to | |
|428602 |rmudgett |push any collected digits into the | |
| | |bridge. | |
|--------+------------+---------------------------------------+----------|
|428604 |rmudgett |test_channel_feature_hooks.c: Fix unit | |
| | |test for DTMF hooks. | |
|--------+------------+---------------------------------------+----------|
|428731 |gtjoseph |res_pjsip_endpoint_identifier_ip: Add | |
| | |'show identify(ies)' cli commands | |
|--------+------------+---------------------------------------+----------|
|428734 |gtjoseph |config: Create | |
| | |ast_variable_find_in_list() | |
|--------+------------+---------------------------------------+----------|
| | |res_pjsip_refer: Fix issue where native| |
|428761 |file |bridge may not occur upon completion of| |
| | |a transfer. | |
|--------+------------+---------------------------------------+----------|
|428815 |mjordan |tests/test_stasis: Resolve compilation | |
| | |issues from Asterisk 12 merge | |
|--------+------------+---------------------------------------+----------|
|428837 |gtjoseph |CHANGES: Add item for new 'pjsip show | |
| | |identif(y|ies) commands | |
|--------+------------+---------------------------------------+----------|
| | |tests/test_cel: Fix CEL unit test | |
|428892 |mjordan |failures caused by attended transfer | |
| | |changes | |
|--------+------------+---------------------------------------+----------|
| | |tests/test_cel: Add | |
|428919 |mjordan |test_cel_attended_transfer_bridges_link| |
| | |to racey tests | |
|--------+------------+---------------------------------------+----------|
|428946 |mjordan |main/test: Fix race condition between | |
| | |AMI topic and Test Suite topic | |
|--------+------------+---------------------------------------+----------|
|428973 |mjordan |main/test: Fix compilation issue on | |
| | |32-bit systems | |
|--------+------------+---------------------------------------+----------|
|429000 |gtjoseph |sorcery: Add additional observer | |
| | |capabilities. | |
|--------+------------+---------------------------------------+----------|
|429091 |mjordan |AMI/ARI: Update version to 2.6.0/1.6.0 | |
| | |respectively for new features | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
CHANGES | 43
Makefile | 4
Makefile.rules | 18
UPGRADE.txt | 25
addons/chan_mobile.c | 2
apps/app_agent_pool.c | 39
apps/app_confbridge.c | 4
apps/app_meetme.c | 35
apps/app_queue.c | 15
apps/app_record.c | 9
apps/app_voicemail.c | 45
apps/confbridge/conf_state_multi_marked.c | 27
bridges/bridge_builtin_features.c | 4
bridges/bridge_native_rtp.c | 10
build_tools/menuselect-deps.in | 1
cel/cel_odbc.c | 13
channels/chan_console.c | 64 -
channels/chan_dahdi.c | 2
channels/chan_iax2.c | 2
channels/chan_mgcp.c | 7
channels/chan_motif.c | 6
channels/chan_phone.c | 2
channels/chan_pjsip.c | 20
channels/chan_sip.c | 113 +-
channels/chan_skinny.c | 2
channels/chan_unistim.c | 8
channels/sig_pri.c | 2
channels/sip/include/reqresp_parser.h | 5
channels/sip/reqresp_parser.c | 6
channels/sip/security_events.c | 2
configs/samples/cdr.conf.sample | 21
configs/samples/extensions.conf.sample | 2
configs/samples/features.conf.sample | 4
configs/samples/manager.conf.sample | 2
configs/samples/pjsip.conf.sample | 13
configs/samples/stasis.conf.sample | 10
configure.ac | 3
contrib/Makefile | 29
contrib/ast-db-manage/config/versions/eb88a14f2a_add_media_encryption_optimistic_to_pjsip.py | 31
funcs/func_cdr.c | 3
funcs/func_db.c | 2
funcs/func_talkdetect.c | 1
include/asterisk/autoconfig.h.in | 3
include/asterisk/bridge_channel.h | 25
include/asterisk/channel.h | 11
include/asterisk/config.h | 14
include/asterisk/features_config.h | 6
include/asterisk/manager.h | 2
include/asterisk/res_pjsip.h | 5
include/asterisk/res_pjsip_session.h | 2
include/asterisk/sorcery.h | 176 +++
include/asterisk/stasis.h | 36
include/asterisk/stasis_app.h | 5
include/asterisk/stasis_bridges.h | 213 ++--
include/asterisk/stasis_channels.h | 14
include/asterisk/stasis_internal.h | 7
include/asterisk/stasis_message_router.h | 16
include/asterisk/stringfields.h | 47 -
include/asterisk/test.h | 4
include/asterisk/utils.h | 10
main/abstract_jb.c | 15
main/acl.c | 2
main/app.c | 18
main/audiohook.c | 2
main/bridge.c | 283 +-----
main/bridge_basic.c | 185 +---
main/bridge_channel.c | 359 +++++--
main/cdr.c | 22
main/cel.c | 4
main/channel.c | 6
main/channel_internal_api.c | 9
main/config.c | 32
main/endpoints.c | 2
main/features.c | 1
main/features_config.c | 34
main/file.c | 4
main/manager.c | 185 ++--
main/manager_channels.c | 20
main/pbx.c | 26
main/rtp_engine.c | 2
main/sorcery.c | 365 ++++++-
main/stasis.c | 152 ++-
main/stasis_bridges.c | 264 ++---
main/stasis_cache.c | 2
main/stasis_channels.c | 10
main/stasis_message_router.c | 22
main/stun.c | 11
main/test.c | 50 -
main/utils.c | 46
makeopts.in | 3
pbx/pbx_config.c | 31
pbx/pbx_loopback.c | 19
res/ari/ari_model_validators.c | 85 +
res/ari/ari_model_validators.h | 23
res/parking/parking_applications.c | 2
res/parking/parking_bridge_features.c | 2
res/parking/parking_tests.c | 2
res/res_agi.c | 4
res/res_calendar.c | 2
res/res_fax.c | 25
res/res_hep.c | 1
res/res_http_websocket.c | 18
res/res_monitor.c | 2
res/res_phoneprov.c | 6
res/res_pjsip.c | 35
res/res_pjsip/config_system.c | 8
res/res_pjsip/pjsip_cli.c | 5
res/res_pjsip/pjsip_configuration.c | 9
res/res_pjsip/pjsip_distributor.c | 28
res/res_pjsip/pjsip_options.c | 15
res/res_pjsip_acl.c | 7
res/res_pjsip_endpoint_identifier_ip.c | 83 +
res/res_pjsip_exten_state.c | 4
res/res_pjsip_multihomed.c | 8
res/res_pjsip_mwi.c | 2
res/res_pjsip_outbound_registration.c | 461 ++++++----
res/res_pjsip_phoneprov_provider.c | 9
res/res_pjsip_pubsub.c | 24
res/res_pjsip_refer.c | 63 +
res/res_pjsip_sdp_rtp.c | 105 +-
res/res_pjsip_session.c | 42
res/res_srtp.c | 1
res/res_stasis.c | 314 +++---
res/res_stasis_device_state.c | 2
res/res_xmpp.c | 2
res/stasis/app.c | 59 -
res/stasis/app.h | 7
res/stasis/stasis_bridge.c | 6
rest-api/api-docs/applications.json | 2
rest-api/api-docs/asterisk.json | 2
rest-api/api-docs/bridges.json | 2
rest-api/api-docs/channels.json | 2
rest-api/api-docs/deviceStates.json | 2
rest-api/api-docs/endpoints.json | 2
rest-api/api-docs/events.json | 16
rest-api/api-docs/mailboxes.json | 2
rest-api/api-docs/playbacks.json | 2
rest-api/api-docs/recordings.json | 2
rest-api/api-docs/sounds.json | 2
rest-api/resources.json | 2
tests/test_cel.c | 72 -
tests/test_channel_feature_hooks.c | 31
tests/test_sorcery.c | 355 +++++++
tests/test_stasis.c | 310 ++++++
tests/test_strings.c | 66 +
145 files changed, 4106 insertions(+), 1675 deletions(-)
----------------------------------------------------------------------
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