[asterisk-commits] bebuild: tag 13.1.0-rc1 r429108 - /tags/13.1.0-rc1/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Dec 8 11:18:33 CST 2014
Author: bebuild
Date: Mon Dec 8 11:18:28 2014
New Revision: 429108
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=429108
Log:
Importing release summary for 13.1.0-rc1 release.
Added:
tags/13.1.0-rc1/asterisk-13.1.0-rc1-summary.html (with props)
tags/13.1.0-rc1/asterisk-13.1.0-rc1-summary.txt (with props)
Added: tags/13.1.0-rc1/asterisk-13.1.0-rc1-summary.html
URL: http://svnview.digium.com/svn/asterisk/tags/13.1.0-rc1/asterisk-13.1.0-rc1-summary.html?view=auto&rev=429108
==============================================================================
--- tags/13.1.0-rc1/asterisk-13.1.0-rc1-summary.html (added)
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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">
+<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"><asteriskteam at digium.com></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>
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