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Importing release summary for 13.3.0-rc1 release.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/13.3.0-rc1@433313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-13.3.0-rc1</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-13.3.0-rc1</h3>
<h3 align="center">Date: 2015-03-23</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-13.2.0.</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>
25 rmudgett<br/>
17 mjordan<br/>
16 jcolp<br/>
9 kharwell<br/>
7 coreyfarrell<br/>
4 file<br/>
4 Graham Barnett<br/>
4 sgriepentrog<br/>
3 gtjoseph<br/>
2 oej<br/>
2 snuffy<br/>
1 Alexander Traud<br/>
1 Ben Merrills<br/>
1 Diederik de Groot<br/>
1 dlee<br/>
1 Ed Hynan<br/>
1 ibercom<br/>
1 Javier Acosta<br/>
1 jrose<br/>
1 Makoto Dei<br/>
1 newtonr<br/>
1 Richard Miller<br/>
</td>
<td>
2 Graham Barnett<br/>
2 snuffy<br/>
1 JoshE<br/>
1 mjordan<br/>
</td>
<td>
7 coreyfarrell<br/>
5 mjordan<br/>
5 rnewton<br/>
4 jcolp<br/>
3 GrahamJB<br/>
3 kharwell<br/>
3 rmudgett<br/>
2 n8ideas<br/>
2 snuffy<br/>
2 zconkle<br/>
1 anatoli<br/>
1 asanders<br/>
1 atis<br/>
1 bford<br/>
1 cbbs70a<br/>
1 DarkS<br/>
1 dcabot<br/>
1 Demon<br/>
1 dhubbard<br/>
1 dkdegroot<br/>
1 Ed<br/>
1 falves11<br/>
1 feyfre<br/>
1 ibercom<br/>
1 jbigelow<br/>
1 jputnam<br/>
1 kenner<br/>
1 klaus3000<br/>
1 makoto<br/>
1 mhoskins<br/>
1 mmichelson<br/>
1 pnlarsson<br/>
1 roeften<br/>
1 rossbeer<br/>
1 simmcomm<br/>
1 skrusty<br/>
1 smurfix<br/>
1 StefanEng86<br/>
1 traud<br/>
1 twilson<br/>
1 ulogic<br/>
1 yurakocyuba<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: Applications/app_amd</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19470">ASTERISK-19470</a>: Documentation on app_amd is incorrect<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432920">432920</a><br/>
Reporter: cbbs70a<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Applications/app_chanspy</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24828">ASTERISK-24828</a>: Fix Frame Leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432363">432363</a><br/>
Reporter: kharwell<br/>
Coders: kharwell<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=432118">432118</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Applications/app_transfer</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24015">ASTERISK-24015</a>: app_transfer fails with PJSIP channels<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431717">431717</a><br/>
Reporter: falves11<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Applications/app_voicemail</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24790">ASTERISK-24790</a>: Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432099">432099</a><br/>
Reporter: GrahamJB<br/>
Coders: Graham Barnett<br/>
<br/>
<h3>Category: Applications/app_voicemail/IMAP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24786">ASTERISK-24786</a>: [patch] - Asterisk terminates when playing a voicemail stored in LDAP<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432696">432696</a><br/>
Reporter: GrahamJB<br/>
Testers: Graham Barnett<br/>
Coders: Graham Barnett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24787">ASTERISK-24787</a>: [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432013">432013</a><br/>
Reporter: GrahamJB<br/>
Coders: Graham Barnett<br/>
<br/>
<h3>Category: Bridges/bridge_softmix</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24797">ASTERISK-24797</a>: bridge_softmix: G.729 codec license held<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432175">432175</a><br/>
Reporter: kharwell<br/>
Coders: kharwell<br/>
<br/>
<h3>Category: Channels/chan_dahdi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24689">ASTERISK-24689</a>: Segfault on hangup after outgoing PRI-Euroisdn call<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431974">431974</a><br/>
Reporter: simmcomm<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24825">ASTERISK-24825</a>: Caller ID not recognized using Centrex/Distinctive dialing<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432534">432534</a><br/>
Reporter: rmudgett<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Channels/chan_iax2</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24451">ASTERISK-24451</a>: chan_iax2: reference leak in sched_delay_remove<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431917">431917</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Channels/chan_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24771">ASTERISK-24771</a>: ${CHANNEL(pjsip)} - segfault<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431751">431751</a><br/>
Reporter: pnlarsson<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Channels/chan_sip/DatabaseSupport</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24772">ASTERISK-24772</a>: ODBC error in realtime sippeers when device unregisters under MariaDB<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431674">431674</a><br/>
Reporter: ulogic<br/>
Coders: Richard Miller<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-15434">ASTERISK-15434</a>: [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432321">432321</a><br/>
Reporter: makoto<br/>
Testers: mjordan<br/>
Coders: Makoto Dei<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23214">ASTERISK-23214</a>: chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432278">432278</a><br/>
Reporter: rnewton<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24800">ASTERISK-24800</a>: Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432199">432199</a><br/>
Reporter: n8ideas<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24838">ASTERISK-24838</a>: chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432528">432528</a><br/>
Reporter: rmudgett<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24876">ASTERISK-24876</a>: Investigate reference leaks from tests/channels/local/local_optimize_away<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433113">433113</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24882">ASTERISK-24882</a>: chan_sip: Improve usage of REF_DEBUG<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433115">433115</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Channels/chan_sip/NewFeature</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17899">ASTERISK-17899</a>: [patch] Adds a 'ignorecryptolifetime' (Ignore Crypto Lifetime) option to sip.conf for SRTP keys specifying optional 'lifetime'<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432258">432258</a><br/>
Reporter: dhubbard<br/>
Coders: oej<br/>
<br/>
<h3>Category: Channels/chan_sip/SRTP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-17721">ASTERISK-17721</a>: Incoming SRTP calls that specify a key lifetime fail<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432258">432258</a><br/>
Reporter: twilson<br/>
Coders: oej<br/>
<br/>
<h3>Category: Channels/chan_sip/TCP-TLS</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24799">ASTERISK-24799</a>: [patch] make fails with undefined reference to SSLv3_client_method<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431937">431937</a><br/>
Reporter: traud<br/>
Coders: Alexander Traud<br/>
<br/>
<h3>Category: Contrib/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24632">ASTERISK-24632</a>: install_prereq script installs pjproject without IPv6 support<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431843">431843</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/Bridging</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24752">ASTERISK-24752</a>: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431692">431692</a><br/>
Reporter: rmudgett<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Core/BuildSystem</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18105">ASTERISK-18105</a>: most of asterisk modules are unbuildable in cygwin environment<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432342">432342</a><br/>
Reporter: feyfre<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-20850">ASTERISK-20850</a>: [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432808">432808</a><br/>
Reporter: dkdegroot<br/>
Coders: Diederik de Groot<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24880">ASTERISK-24880</a>: [patch]Compilation under OpenBSD <br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433247">433247</a><br/>
Reporter: snuffy<br/>
Testers: snuffy<br/>
Coders: snuffy<br/>
<br/>
<h3>Category: Core/Channels</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-21038">ASTERISK-21038</a>: Bad command completion of "core set debug channel"<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432945">432945</a><br/>
Reporter: kenner<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24828">ASTERISK-24828</a>: Fix Frame Leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432363">432363</a><br/>
Reporter: kharwell<br/>
Coders: kharwell<br/>
<br/>
<h3>Category: Core/CodecInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-16779">ASTERISK-16779</a>: Cannot disallow unknown format ''<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432971">432971</a><br/>
Reporter: atis<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24796">ASTERISK-24796</a>: Codecs and bucket schema's prevent module unload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432059">432059</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24479">ASTERISK-24479</a>: Enable REF_DEBUG for module references<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431663">431663</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24739">ASTERISK-24739</a>: [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432693">432693</a><br/>
Reporter: Ed<br/>
Coders: Ed Hynan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24740">ASTERISK-24740</a>: [patch]Segmentation fault on aoc-e event<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431974">431974</a><br/>
Reporter: roeften<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24752">ASTERISK-24752</a>: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431692">431692</a><br/>
Reporter: rmudgett<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24796">ASTERISK-24796</a>: Codecs and bucket schema's prevent module unload<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432059">432059</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24814">ASTERISK-24814</a>: asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432055">432055</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24879">ASTERISK-24879</a>: [patch]Compilation fails due to 64bit time under OpenBSD<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433269">433269</a><br/>
Reporter: snuffy<br/>
Testers: snuffy<br/>
Coders: snuffy<br/>
<br/>
<h3>Category: Core/HTTP</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24724">ASTERISK-24724</a>: 'httpstatus' Web Page Produces Incomplete HTML<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432079">432079</a><br/>
Reporter: asanders<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/Logging</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24817">ASTERISK-24817</a>: init_logger_chain: unreachable code block<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433126">433126</a><br/>
Reporter: coreyfarrell<br/>
Coders: coreyfarrell<br/>
<br/>
<h3>Category: Core/ManagerInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22670">ASTERISK-22670</a>: Asterisk crashes when processing ISDN AoC Events<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431974">431974</a><br/>
Reporter: klaus3000<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Core/Sorcery</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24612">ASTERISK-24612</a>: res_pjsip: No information if a required sorcery wizard is not loaded<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431771">431771</a><br/>
Reporter: jcolp<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Documentation</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24085">ASTERISK-24085</a>: Documentation - We should remove or further document the 'contact' section in pjsip.conf<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431860">431860</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Functions/func_curl</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-18708">ASTERISK-18708</a>: func_curl hangs channel under load<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432949">432949</a><br/>
Reporter: dcabot<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_agi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23390">ASTERISK-23390</a>: NewExten Event with application AGI shows up before and after AGI runs<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432696">432696</a><br/>
Reporter: bford<br/>
Testers: Graham Barnett<br/>
Coders: Graham Barnett<br/>
<br/>
<h3>Category: Resources/res_ari</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24812">ASTERISK-24812</a>: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432195">432195</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_ari_channels</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24677">ASTERISK-24677</a>: ARI GET variable on channel provides unhelpful response on non-existent variable<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432154">432154</a><br/>
Reporter: jcolp<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24703">ASTERISK-24703</a>: ARI: Add the ability to "transfer" (redirect) a channel<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431717">431717</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24812">ASTERISK-24812</a>: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432195">432195</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_config_odbc</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24808">ASTERISK-24808</a>: res_config_odbc: Improper escaping of backslashes occurs with MySQL<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432721">432721</a><br/>
Reporter: DarkS<br/>
Coders: Javier Acosta<br/>
<br/>
<h3>Category: Resources/res_format_attr_h264</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24616">ASTERISK-24616</a>: Crash in res_format_attr_h264 due to invalid string copy<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431521">431521</a><br/>
Reporter: yurakocyuba<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_odbc</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24742">ASTERISK-24742</a>: [patch] Fix ast_odbc_find_table function in res_odbc<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431618">431618</a><br/>
Reporter: ibercom<br/>
Coders: ibercom<br/>
<br/>
<h3>Category: Resources/res_pjsip</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=432118">432118</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24685">ASTERISK-24685</a>: "pjsip show version" CLI command<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431824">431824</a><br/>
Reporter: jcolp<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24727">ASTERISK-24727</a>: PJSIP: Crash experienced during multi-Asterisk transfer scenario.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431734">431734</a><br/>
Reporter: mmichelson<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24741">ASTERISK-24741</a>: dtls_handler causes Asterisk to crash<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431698">431698</a><br/>
Reporter: zconkle<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24748">ASTERISK-24748</a>: res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431538">431538</a><br/>
Reporter: jcolp<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24755">ASTERISK-24755</a>: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432668">432668</a><br/>
Reporter: jbigelow<br/>
Coders: rmudgett<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24840">ASTERISK-24840</a>: res_pjsip: conflicting endpoint identifiers<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432638">432638</a><br/>
Reporter: kharwell<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24872">ASTERISK-24872</a>: [patch] AMI PJSIPShowEndpoint closes AMI connection on error<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432894">432894</a><br/>
Reporter: Demon<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Resources/res_pjsip_exten_state</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24716">ASTERISK-24716</a>: Improve pjsip log messages for presence subscription failure<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431754">431754</a><br/>
Reporter: rnewton<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_pjsip_publish_asterisk</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24811">ASTERISK-24811</a>: asterisk-publication sorcery object does not use realtime<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432033">432033</a><br/>
Reporter: mhoskins<br/>
Coders: gtjoseph<br/>
<br/>
<h3>Category: Resources/res_pjsip_refer</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24700">ASTERISK-24700</a>: CRASH: NULL channel is being passed to ast_bridge_transfer_attended()<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431898">431898</a><br/>
Reporter: zconkle<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Resources/res_pjsip_registrar</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24785">ASTERISK-24785</a>: 'Expires' header missing from 200 OK on REGISTER<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432136">432136</a><br/>
Reporter: rossbeer<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_pjsip_sdp_rtp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24769">ASTERISK-24769</a>: res_pjsip_sdp_rtp: Local ICE candidates leaked<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431600">431600</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_rtp_asterisk</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24791">ASTERISK-24791</a>: Crash in ast_rtcp_write_report<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431879">431879</a><br/>
Reporter: n8ideas<br/>
Testers: JoshE<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_stasis</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24701">ASTERISK-24701</a>: Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431670">431670</a><br/>
Reporter: mjordan<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24701">ASTERISK-24701</a>: Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431693">431693</a><br/>
Reporter: mjordan<br/>
Coders: kharwell<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24755">ASTERISK-24755</a>: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432668">432668</a><br/>
Reporter: jbigelow<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Resources/res_timing_pthread</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24768">ASTERISK-24768</a>: res_timing_pthread: file descriptor leak<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431807">431807</a><br/>
Reporter: smurfix<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: pjproject/pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24807">ASTERISK-24807</a>: Missing mandatory field Max-Forwards<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432766">432766</a><br/>
Reporter: anatoli<br/>
Coders: rmudgett<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=431555">431555</a></td><td>file</td><td>res_pjsip_keepalive: Don't crash if PJSIP module is not loaded.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431583">431583</a></td><td>sgriepentrog</td><td>various: cleanup issues found during leak hunt</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431643">431643</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: Add ability to auto-create hints.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431752">431752</a></td><td>file</td><td>'information' ends with an 'n'.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431789">431789</a></td><td>mjordan</td><td>apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431841">431841</a></td><td>file</td><td>res_sorcery_config: Improve object lookup times.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431956">431956</a></td><td>rmudgett</td><td>res_pjsip_refer: Handle INVITE with Replaces failure after answer.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=431993">431993</a></td><td>rmudgett</td><td>chan_dahdi: Remove some dead code.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432034">432034</a></td><td>rmudgett</td><td>chan_dahdi/sig_analog: Put log message strings on one line.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432237">432237</a></td><td>dlee</td><td>Increase WebSocket frame size and improve large read handling</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432281">432281</a></td><td>mjordan</td><td>configure: Promote SQLite3 "not installed" warning to error</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432301">432301</a></td><td>newtonr</td><td>configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432385">432385</a></td><td>sgriepentrog</td><td>Dial API: add self destruct option when complete</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432423">432423</a></td><td>mjordan</td><td>res/res_pjsip_sdp_rtp: Revert portion of r432195</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432453">432453</a></td><td>mjordan</td><td>translate: Prevent invalid memory accesses on fast shutdown</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432485">432485</a></td><td>gtjoseph</td><td>app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432556">432556</a></td><td>jrose</td><td>app: Add functions to swap voicemail function table for testing purposes</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432574">432574</a></td><td>rmudgett</td><td>res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432594">432594</a></td><td>rmudgett</td><td>res_pjsip_refer: Make safely get the context for a blind transfer.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432742">432742</a></td><td>file</td><td>core: Don't create snapshots with locks.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432764">432764</a></td><td>rmudgett</td><td>res_pjsip: Fixed invalid empty Server and User-Agent SIP headers.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432787">432787</a></td><td>rmudgett</td><td>res_pjsip: Move internal init/destroy prototypes to private header file.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432811">432811</a></td><td>mjordan</td><td>main/audiohook: Update internal sample rate on reads</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432868">432868</a></td><td>kharwell</td><td>Revert - res_pjsip: Allow configuration of endpoint identifier query order</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432892">432892</a></td><td>rmudgett</td><td>chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=432938">432938</a></td><td>mjordan</td><td>FILE: fix retrieval of file contents when offset is specified</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433005">433005</a></td><td>rmudgett</td><td>res_pjsip: Add reason comment.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433028">433028</a></td><td>kharwell</td><td>res_pjsip: Allow configuration of endpoint identifier query order</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433031">433031</a></td><td>kharwell</td><td>res_pjsip: Allow configuration of endpoint identifier query order</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433057">433057</a></td><td>rmudgett</td><td>Audit ast_sockaddr_resolve() usage for memory leaks.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433060">433060</a></td><td>sgriepentrog</td><td>core: Introduce chaos into memory allocations</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433064">433064</a></td><td>sgriepentrog</td><td>Various: bugfixes found via chaos</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433088">433088</a></td><td>rmudgett</td><td>res_pjsip_session: Fix off-nominal extra unref of session.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433174">433174</a></td><td>mjordan</td><td>funcs/func_env: Fix regression caused in FILE read operation</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433199">433199</a></td><td>rmudgett</td><td>res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=433222">433222</a></td><td>rmudgett</td><td>Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks.</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 | 25
Makefile | 4
UPGRADE.txt | 9
apps/Makefile | 2
apps/app_amd.c | 3
apps/app_chanspy.c | 3
apps/app_confbridge.c | 6
apps/app_externalivr.c | 2
apps/app_mixmonitor.c | 7
apps/app_queue.c | 26
apps/app_voicemail.c | 46
bridges/bridge_builtin_features.c | 8
bridges/bridge_softmix.c | 37
build_tools/cflags.xml | 4
channels/Makefile | 2
channels/chan_dahdi.c | 124 +-
channels/chan_dahdi.h | 2
channels/chan_iax2.c | 4
channels/chan_pjsip.c | 128 +-
channels/chan_sip.c | 179 +--
channels/pjsip/dialplan_functions.c | 5
channels/sig_analog.c | 192 +--
channels/sig_analog.h | 1
channels/sip/include/dialog.h | 15
channels/sip/include/sip.h | 11
configs/basic-pbx/README | 15
configs/basic-pbx/asterisk.conf | 26
configs/basic-pbx/extensions.conf | 58 +
configs/basic-pbx/indications.conf | 19
configs/basic-pbx/logger.conf | 9
configs/basic-pbx/modules.conf | 102 ++
configs/basic-pbx/musiconhold.conf | 5
configs/basic-pbx/pjsip.conf | 287 +++++
configs/basic-pbx/voicemail.conf | 23
configs/samples/amd.conf.sample | 1
configs/samples/pjsip.conf.sample | 15
configs/samples/pjsip_wizard.conf.sample | 22
configure.ac | 49 -
contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py | 21
contrib/scripts/install_prereq | 2
funcs/func_cdr.c | 6
funcs/func_curl.c | 2
funcs/func_env.c | 2
include/asterisk.h | 36
include/asterisk/app.h | 16
include/asterisk/channel.h | 19
include/asterisk/config.h | 5
include/asterisk/dial.h | 1
include/asterisk/inline_api.h | 12
include/asterisk/json.h | 18
include/asterisk/lock.h | 2
include/asterisk/module.h | 27
include/asterisk/res_pjsip.h | 119 --
include/asterisk/res_pjsip_session.h | 14
include/asterisk/sched.h | 11
include/asterisk/stasis_app.h | 11
include/asterisk/utils.h | 67 +
main/Makefile | 1
main/aoc.c | 88 +
main/app.c | 51 +
main/asterisk.c | 255 +++--
main/audiohook.c | 15
main/bridge.c | 14
main/bucket.c | 2
main/cdr.c | 8
main/channel.c | 40
main/cli.c | 4
main/codec.c | 4
main/codec_builtin.c | 2
main/config.c | 17
main/dial.c | 8
main/endpoints.c | 8
main/format_cap.c | 4
main/http.c | 9
main/json.c | 29
main/loader.c | 79 +
main/logger.c | 31
main/manager.c | 7
main/netsock2.c | 4
main/rtp_engine.c | 2
main/sched.c | 20
main/sdp_srtp.c | 109 +-
main/sorcery.c | 11
main/stasis_bridges.c | 3
main/stasis_channels.c | 6
main/stdtime/localtime.c | 279 ++++-
main/tcptls.c | 5
main/translate.c | 4
main/utils.c | 4
main/xmldoc.c | 9
makeopts.in | 2
res/ari/ari_websockets.c | 14
res/ari/resource_channels.c | 136 ++
res/ari/resource_channels.h | 26
res/ari/resource_endpoints.c | 61 -
res/res_ari_channels.c | 115 ++
res/res_ari_endpoints.c | 1
res/res_config_odbc.c | 8
res/res_format_attr_h264.c | 4
res/res_http_websocket.c | 43
res/res_odbc.c | 11
res/res_pjsip.c | 150 ++-
res/res_pjsip/config_domain_aliases.c | 1
res/res_pjsip/config_global.c | 152 ++-
res/res_pjsip/include/res_pjsip_private.h | 164 +++
res/res_pjsip/pjsip_cli.c | 25
res/res_pjsip/pjsip_configuration.c | 14
res/res_pjsip/pjsip_global_headers.c | 14
res/res_pjsip/pjsip_options.c | 23
res/res_pjsip_acl.c | 3
res/res_pjsip_caller_id.c | 53 -
res/res_pjsip_config_wizard.c | 224 ++++
res/res_pjsip_endpoint_identifier_anonymous.c | 2
res/res_pjsip_endpoint_identifier_ip.c | 2
res/res_pjsip_endpoint_identifier_user.c | 2
res/res_pjsip_exten_state.c | 3
res/res_pjsip_keepalive.c | 2
res/res_pjsip_messaging.c | 24
res/res_pjsip_multihomed.c | 5
res/res_pjsip_nat.c | 12
res/res_pjsip_publish_asterisk.c | 1
res/res_pjsip_pubsub.c | 2
res/res_pjsip_refer.c | 263 +++--
res/res_pjsip_registrar.c | 8
res/res_pjsip_sdp_rtp.c | 22
res/res_pjsip_send_to_voicemail.c | 10
res/res_pjsip_session.c | 488 +++++++---
res/res_pjsip_session.exports.in | 1
res/res_pjsip_t38.c | 4
res/res_pjsip_transport_websocket.c | 2
res/res_rtp_asterisk.c | 20
res/res_sorcery_config.c | 46
res/res_timing_pthread.c | 3
res/stasis/control.c | 32
rest-api/api-docs/channels.json | 53 +
rest-api/api-docs/endpoints.json | 4
tests/test_func_file.c | 6
137 files changed, 4102 insertions(+), 1158 deletions(-)
</pre><br/>
<hr/>
</body>
</html>

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asterisk-13.3.0-rc1-summary.txt Normal file
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@@ -0,0 +1,877 @@
Release Summary
asterisk-13.3.0-rc1
Date: 2015-03-23
<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-13.2.0.
----------------------------------------------------------------------
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
25 rmudgett 2 Graham Barnett 7 coreyfarrell
17 mjordan 2 snuffy 5 mjordan
16 jcolp 1 JoshE 5 rnewton
9 kharwell 1 mjordan 4 jcolp
7 coreyfarrell 3 GrahamJB
4 file 3 kharwell
4 Graham Barnett 3 rmudgett
4 sgriepentrog 2 n8ideas
3 gtjoseph 2 snuffy
2 oej 2 zconkle
2 snuffy 1 anatoli
1 Alexander Traud 1 asanders
1 Ben Merrills 1 atis
1 Diederik de Groot 1 bford
1 dlee 1 cbbs70a
1 Ed Hynan 1 DarkS
1 ibercom 1 dcabot
1 Javier Acosta 1 Demon
1 jrose 1 dhubbard
1 Makoto Dei 1 dkdegroot
1 newtonr 1 Ed
1 Richard Miller 1 falves11
1 feyfre
1 ibercom
1 jbigelow
1 jputnam
1 kenner
1 klaus3000
1 makoto
1 mhoskins
1 mmichelson
1 pnlarsson
1 roeften
1 rossbeer
1 simmcomm
1 skrusty
1 smurfix
1 StefanEng86
1 traud
1 twilson
1 ulogic
1 yurakocyuba
----------------------------------------------------------------------
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: Applications/app_amd
ASTERISK-19470: Documentation on app_amd is incorrect
Revision: 432920
Reporter: cbbs70a
Coders: mjordan
Category: Applications/app_chanspy
ASTERISK-24828: Fix Frame Leaks
Revision: 432363
Reporter: kharwell
Coders: kharwell
Category: Applications/app_dial
ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Revision: 432118
Reporter: rnewton
Coders: jcolp
Category: Applications/app_transfer
ASTERISK-24015: app_transfer fails with PJSIP channels
Revision: 431717
Reporter: falves11
Coders: mjordan
Category: Applications/app_voicemail
ASTERISK-24790: Reduce spurious noise in logs from voicemail - Couldn't
find mailbox %s in context
Revision: 432099
Reporter: GrahamJB
Coders: Graham Barnett
Category: Applications/app_voicemail/IMAP
ASTERISK-24786: [patch] - Asterisk terminates when playing a voicemail
stored in LDAP
Revision: 432696
Reporter: GrahamJB
Testers: Graham Barnett
Coders: Graham Barnett
ASTERISK-24787: [patch] - Microsoft exchange incompatibility for playing
back messages stored in IMAP - play_message: No origtime
Revision: 432013
Reporter: GrahamJB
Coders: Graham Barnett
Category: Bridges/bridge_softmix
ASTERISK-24797: bridge_softmix: G.729 codec license held
Revision: 432175
Reporter: kharwell
Coders: kharwell
Category: Channels/chan_dahdi
ASTERISK-24689: Segfault on hangup after outgoing PRI-Euroisdn call
Revision: 431974
Reporter: simmcomm
Coders: rmudgett
ASTERISK-24825: Caller ID not recognized using Centrex/Distinctive dialing
Revision: 432534
Reporter: rmudgett
Coders: rmudgett
Category: Channels/chan_iax2
ASTERISK-24451: chan_iax2: reference leak in sched_delay_remove
Revision: 431917
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Channels/chan_pjsip
ASTERISK-24771: ${CHANNEL(pjsip)} - segfault
Revision: 431751
Reporter: pnlarsson
Coders: jcolp
Category: Channels/chan_sip/DatabaseSupport
ASTERISK-24772: ODBC error in realtime sippeers when device unregisters
under MariaDB
Revision: 431674
Reporter: ulogic
Coders: Richard Miller
Category: Channels/chan_sip/General
ASTERISK-15434: [patch] When ast_pbx_start failed, both an error response
and BYE are sent to the caller
Revision: 432321
Reporter: makoto
Testers: mjordan
Coders: Makoto Dei
ASTERISK-23214: chan_sip WARNING message 'We are requesting SRTP for
audio, but they responded without it' is ambiguous and wrong in some cases
Revision: 432278
Reporter: rnewton
Coders: mjordan
ASTERISK-24800: Crash in __sip_reliable_xmit due to invalid thread ID
being passed to pthread_kill
Revision: 432199
Reporter: n8ideas
Coders: mjordan
ASTERISK-24838: chan_sip: Locking inversion occurs when building a peer
causes a peer poke during request handling
Revision: 432528
Reporter: rmudgett
Coders: rmudgett
ASTERISK-24876: Investigate reference leaks from
tests/channels/local/local_optimize_away
Revision: 433113
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24882: chan_sip: Improve usage of REF_DEBUG
Revision: 433115
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Channels/chan_sip/NewFeature
ASTERISK-17899: [patch] Adds a 'ignorecryptolifetime' (Ignore Crypto
Lifetime) option to sip.conf for SRTP keys specifying optional 'lifetime'
Revision: 432258
Reporter: dhubbard
Coders: oej
Category: Channels/chan_sip/SRTP
ASTERISK-17721: Incoming SRTP calls that specify a key lifetime fail
Revision: 432258
Reporter: twilson
Coders: oej
Category: Channels/chan_sip/TCP-TLS
ASTERISK-24799: [patch] make fails with undefined reference to
SSLv3_client_method
Revision: 431937
Reporter: traud
Coders: Alexander Traud
Category: Contrib/General
ASTERISK-24632: install_prereq script installs pjproject without IPv6
support
Revision: 431843
Reporter: rnewton
Coders: jcolp
Category: Core/Bridging
ASTERISK-24752: Crash in bridge_manager_service_req when bridge is
destroyed by ARI during shutdown
Revision: 431692
Reporter: rmudgett
Coders: rmudgett
Category: Core/BuildSystem
ASTERISK-18105: most of asterisk modules are unbuildable in cygwin
environment
Revision: 432342
Reporter: feyfre
Coders: mjordan
ASTERISK-20850: [patch]Nested functions aren't portable. Adapting RAII_VAR
to use clang/llvm blocks to get the same/similar functionality.
Revision: 432808
Reporter: dkdegroot
Coders: Diederik de Groot
ASTERISK-24880: [patch]Compilation under OpenBSD
Revision: 433247
Reporter: snuffy
Testers: snuffy
Coders: snuffy
Category: Core/Channels
ASTERISK-21038: Bad command completion of "core set debug channel"
Revision: 432945
Reporter: kenner
Coders: jcolp
ASTERISK-24828: Fix Frame Leaks
Revision: 432363
Reporter: kharwell
Coders: kharwell
Category: Core/CodecInterface
ASTERISK-16779: Cannot disallow unknown format ''
Revision: 432971
Reporter: atis
Coders: mjordan
ASTERISK-24796: Codecs and bucket schema's prevent module unload
Revision: 432059
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/General
ASTERISK-24479: Enable REF_DEBUG for module references
Revision: 431663
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24739: [patch] - Out of files -- call fails -- numerous files
with inodes from under /usr/share/zoneinfo, mostly posixrules
Revision: 432693
Reporter: Ed
Coders: Ed Hynan
ASTERISK-24740: [patch]Segmentation fault on aoc-e event
Revision: 431974
Reporter: roeften
Coders: rmudgett
ASTERISK-24752: Crash in bridge_manager_service_req when bridge is
destroyed by ARI during shutdown
Revision: 431692
Reporter: rmudgett
Coders: rmudgett
ASTERISK-24796: Codecs and bucket schema's prevent module unload
Revision: 432059
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24814: asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64
bit integers
Revision: 432055
Reporter: coreyfarrell
Coders: coreyfarrell
ASTERISK-24879: [patch]Compilation fails due to 64bit time under OpenBSD
Revision: 433269
Reporter: snuffy
Testers: snuffy
Coders: snuffy
Category: Core/HTTP
ASTERISK-24724: 'httpstatus' Web Page Produces Incomplete HTML
Revision: 432079
Reporter: asanders
Coders: jcolp
Category: Core/Logging
ASTERISK-24817: init_logger_chain: unreachable code block
Revision: 433126
Reporter: coreyfarrell
Coders: coreyfarrell
Category: Core/ManagerInterface
ASTERISK-22670: Asterisk crashes when processing ISDN AoC Events
Revision: 431974
Reporter: klaus3000
Coders: rmudgett
Category: Core/Sorcery
ASTERISK-24612: res_pjsip: No information if a required sorcery wizard is
not loaded
Revision: 431771
Reporter: jcolp
Coders: jcolp
Category: Documentation
ASTERISK-24085: Documentation - We should remove or further document the
'contact' section in pjsip.conf
Revision: 431860
Reporter: rnewton
Coders: jcolp
Category: Functions/func_curl
ASTERISK-18708: func_curl hangs channel under load
Revision: 432949
Reporter: dcabot
Coders: jcolp
Category: Resources/res_agi
ASTERISK-23390: NewExten Event with application AGI shows up before and
after AGI runs
Revision: 432696
Reporter: bford
Testers: Graham Barnett
Coders: Graham Barnett
Category: Resources/res_ari
ASTERISK-24812: ARI: Creating channels through /channels resource always
uses SLIN, which results in unneeded transcoding
Revision: 432195
Reporter: mjordan
Coders: mjordan
Category: Resources/res_ari_channels
ASTERISK-24677: ARI GET variable on channel provides unhelpful response on
non-existent variable
Revision: 432154
Reporter: jcolp
Coders: jcolp
ASTERISK-24703: ARI: Add the ability to "transfer" (redirect) a channel
Revision: 431717
Reporter: mjordan
Coders: mjordan
ASTERISK-24812: ARI: Creating channels through /channels resource always
uses SLIN, which results in unneeded transcoding
Revision: 432195
Reporter: mjordan
Coders: mjordan
Category: Resources/res_config_odbc
ASTERISK-24808: res_config_odbc: Improper escaping of backslashes occurs
with MySQL
Revision: 432721
Reporter: DarkS
Coders: Javier Acosta
Category: Resources/res_format_attr_h264
ASTERISK-24616: Crash in res_format_attr_h264 due to invalid string copy
Revision: 431521
Reporter: yurakocyuba
Coders: jcolp
Category: Resources/res_odbc
ASTERISK-24742: [patch] Fix ast_odbc_find_table function in res_odbc
Revision: 431618
Reporter: ibercom
Coders: ibercom
Category: Resources/res_pjsip
ASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Revision: 432118
Reporter: rnewton
Coders: jcolp
ASTERISK-24685: "pjsip show version" CLI command
Revision: 431824
Reporter: jcolp
Coders: jcolp
ASTERISK-24727: PJSIP: Crash experienced during multi-Asterisk transfer
scenario.
Revision: 431734
Reporter: mmichelson
Coders: rmudgett
ASTERISK-24741: dtls_handler causes Asterisk to crash
Revision: 431698
Reporter: zconkle
Coders: kharwell
ASTERISK-24748: res_pjsip: If wizards explicitly configured in
sorcery.conf false ERROR messages may occur
Revision: 431538
Reporter: jcolp
Coders: jcolp
ASTERISK-24755: Asterisk sends unexpected early BYE to transferrer during
attended transfer when using a Stasis bridge
Revision: 432668
Reporter: jbigelow
Coders: rmudgett
ASTERISK-24840: res_pjsip: conflicting endpoint identifiers
Revision: 432638
Reporter: kharwell
Coders: kharwell
ASTERISK-24872: [patch] AMI PJSIPShowEndpoint closes AMI connection on
error
Revision: 432894
Reporter: Demon
Coders: rmudgett
Category: Resources/res_pjsip_exten_state
ASTERISK-24716: Improve pjsip log messages for presence subscription
failure
Revision: 431754
Reporter: rnewton
Coders: jcolp
Category: Resources/res_pjsip_publish_asterisk
ASTERISK-24811: asterisk-publication sorcery object does not use realtime
Revision: 432033
Reporter: mhoskins
Coders: gtjoseph
Category: Resources/res_pjsip_refer
ASTERISK-24700: CRASH: NULL channel is being passed to
ast_bridge_transfer_attended()
Revision: 431898
Reporter: zconkle
Coders: rmudgett
Category: Resources/res_pjsip_registrar
ASTERISK-24785: 'Expires' header missing from 200 OK on REGISTER
Revision: 432136
Reporter: rossbeer
Coders: jcolp
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-24769: res_pjsip_sdp_rtp: Local ICE candidates leaked
Revision: 431600
Reporter: mjordan
Coders: mjordan
Category: Resources/res_rtp_asterisk
ASTERISK-24791: Crash in ast_rtcp_write_report
Revision: 431879
Reporter: n8ideas
Testers: JoshE
Coders: mjordan
Category: Resources/res_stasis
ASTERISK-24701: Stasis: Write timeout on WebSocket fails to fully
disconnect underlying socket, leading to events being dropped with no
additional information
Revision: 431670
Reporter: mjordan
Coders: kharwell
ASTERISK-24701: Stasis: Write timeout on WebSocket fails to fully
disconnect underlying socket, leading to events being dropped with no
additional information
Revision: 431693
Reporter: mjordan
Coders: kharwell
ASTERISK-24755: Asterisk sends unexpected early BYE to transferrer during
attended transfer when using a Stasis bridge
Revision: 432668
Reporter: jbigelow
Coders: rmudgett
Category: Resources/res_timing_pthread
ASTERISK-24768: res_timing_pthread: file descriptor leak
Revision: 431807
Reporter: smurfix
Coders: jcolp
Category: pjproject/pjsip
ASTERISK-24807: Missing mandatory field Max-Forwards
Revision: 432766
Reporter: anatoli
Coders: rmudgett
----------------------------------------------------------------------
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 |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_keepalive: Don't | |
| 431555 | file | crash if PJSIP module is not | |
| | | loaded. | |
|----------+--------------+---------------------------------+------------|
| 431583 | sgriepentrog | various: cleanup issues found | |
| | | during leak hunt | |
|----------+--------------+---------------------------------+------------|
| 431643 | gtjoseph | res_pjsip_config_wizard: Add | |
| | | ability to auto-create hints. | |
|----------+--------------+---------------------------------+------------|
| 431752 | file | 'information' ends with an 'n'. | |
|----------+--------------+---------------------------------+------------|
| | | apps/app_mixmonitor: Move Test | |
| 431789 | mjordan | Event for MIXMONITOR_END to | |
| | | after it finishes | |
|----------+--------------+---------------------------------+------------|
| 431841 | file | res_sorcery_config: Improve | |
| | | object lookup times. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_refer: Handle INVITE | |
| 431956 | rmudgett | with Replaces failure after | |
| | | answer. | |
|----------+--------------+---------------------------------+------------|
| 431993 | rmudgett | chan_dahdi: Remove some dead | |
| | | code. | |
|----------+--------------+---------------------------------+------------|
| 432034 | rmudgett | chan_dahdi/sig_analog: Put log | |
| | | message strings on one line. | |
|----------+--------------+---------------------------------+------------|
| 432237 | dlee | Increase WebSocket frame size | |
| | | and improve large read handling | |
|----------+--------------+---------------------------------+------------|
| 432281 | mjordan | configure: Promote SQLite3 "not | |
| | | installed" warning to error | |
|----------+--------------+---------------------------------+------------|
| | | configs/basic-pbx - Super | |
| 432301 | newtonr | Awesome Company example configs | |
| | | Phase 1, Patch 1 | |
|----------+--------------+---------------------------------+------------|
| 432385 | sgriepentrog | Dial API: add self destruct | |
| | | option when complete | |
|----------+--------------+---------------------------------+------------|
| 432423 | mjordan | res/res_pjsip_sdp_rtp: Revert | |
| | | portion of r432195 | |
|----------+--------------+---------------------------------+------------|
| | | translate: Prevent invalid | |
| 432453 | mjordan | memory accesses on fast | |
| | | shutdown | |
|----------+--------------+---------------------------------+------------|
| | | app_voicemail: Fix compile | |
| 432485 | gtjoseph | breaking in app_voicemail with | |
| | | IMAP_STORAGE. | |
|----------+--------------+---------------------------------+------------|
| | | app: Add functions to swap | |
| 432556 | jrose | voicemail function table for | |
| | | testing purposes | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_refer: Made | |
| 432574 | rmudgett | refer_attended_alloc() not | |
| | | create the ao2 object with a | |
| | | lock. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_refer: Make safely | |
| 432594 | rmudgett | get the context for a blind | |
| | | transfer. | |
|----------+--------------+---------------------------------+------------|
| 432742 | file | core: Don't create snapshots | |
| | | with locks. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip: Fixed invalid empty | |
| 432764 | rmudgett | Server and User-Agent SIP | |
| | | headers. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip: Move internal | |
| 432787 | rmudgett | init/destroy prototypes to | |
| | | private header file. | |
|----------+--------------+---------------------------------+------------|
| 432811 | mjordan | main/audiohook: Update internal | |
| | | sample rate on reads | |
|----------+--------------+---------------------------------+------------|
| | | Revert - res_pjsip: Allow | |
| 432868 | kharwell | configuration of endpoint | |
| | | identifier query order | |
|----------+--------------+---------------------------------+------------|
| | | chan_pjsip/res_pjsip_callerid: | |
| 432892 | rmudgett | Make Party ID handling simpler | |
| | | and consistent. | |
|----------+--------------+---------------------------------+------------|
| | | FILE: fix retrieval of file | |
| 432938 | mjordan | contents when offset is | |
| | | specified | |
|----------+--------------+---------------------------------+------------|
| 433005 | rmudgett | res_pjsip: Add reason comment. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip: Allow configuration | |
| 433028 | kharwell | of endpoint identifier query | |
| | | order | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip: Allow configuration | |
| 433031 | kharwell | of endpoint identifier query | |
| | | order | |
|----------+--------------+---------------------------------+------------|
| 433057 | rmudgett | Audit ast_sockaddr_resolve() | |
| | | usage for memory leaks. | |
|----------+--------------+---------------------------------+------------|
| 433060 | sgriepentrog | core: Introduce chaos into | |
| | | memory allocations | |
|----------+--------------+---------------------------------+------------|
| 433064 | sgriepentrog | Various: bugfixes found via | |
| | | chaos | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_session: Fix | |
| 433088 | rmudgett | off-nominal extra unref of | |
| | | session. | |
|----------+--------------+---------------------------------+------------|
| 433174 | mjordan | funcs/func_env: Fix regression | |
| | | caused in FILE read operation | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_sdp_rtp,sorcery: Fix | |
| 433199 | rmudgett | invalid access and memory leak | |
| | | respectively. | |
|----------+--------------+---------------------------------+------------|
| | | Audit | |
| 433222 | rmudgett | ast_pjsip_rdata_get_endpoint() | |
| | | usage for ref leaks. | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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 | 25
Makefile | 4
UPGRADE.txt | 9
apps/Makefile | 2
apps/app_amd.c | 3
apps/app_chanspy.c | 3
apps/app_confbridge.c | 6
apps/app_externalivr.c | 2
apps/app_mixmonitor.c | 7
apps/app_queue.c | 26
apps/app_voicemail.c | 46
bridges/bridge_builtin_features.c | 8
bridges/bridge_softmix.c | 37
build_tools/cflags.xml | 4
channels/Makefile | 2
channels/chan_dahdi.c | 124 +-
channels/chan_dahdi.h | 2
channels/chan_iax2.c | 4
channels/chan_pjsip.c | 128 +-
channels/chan_sip.c | 179 +--
channels/pjsip/dialplan_functions.c | 5
channels/sig_analog.c | 192 +--
channels/sig_analog.h | 1
channels/sip/include/dialog.h | 15
channels/sip/include/sip.h | 11
configs/basic-pbx/README | 15
configs/basic-pbx/asterisk.conf | 26
configs/basic-pbx/extensions.conf | 58 +
configs/basic-pbx/indications.conf | 19
configs/basic-pbx/logger.conf | 9
configs/basic-pbx/modules.conf | 102 ++
configs/basic-pbx/musiconhold.conf | 5
configs/basic-pbx/pjsip.conf | 287 +++++
configs/basic-pbx/voicemail.conf | 23
configs/samples/amd.conf.sample | 1
configs/samples/pjsip.conf.sample | 15
configs/samples/pjsip_wizard.conf.sample | 22
configure.ac | 49 -
contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py | 21
contrib/scripts/install_prereq | 2
funcs/func_cdr.c | 6
funcs/func_curl.c | 2
funcs/func_env.c | 2
include/asterisk.h | 36
include/asterisk/app.h | 16
include/asterisk/channel.h | 19
include/asterisk/config.h | 5
include/asterisk/dial.h | 1
include/asterisk/inline_api.h | 12
include/asterisk/json.h | 18
include/asterisk/lock.h | 2
include/asterisk/module.h | 27
include/asterisk/res_pjsip.h | 119 --
include/asterisk/res_pjsip_session.h | 14
include/asterisk/sched.h | 11
include/asterisk/stasis_app.h | 11
include/asterisk/utils.h | 67 +
main/Makefile | 1
main/aoc.c | 88 +
main/app.c | 51 +
main/asterisk.c | 255 +++--
main/audiohook.c | 15
main/bridge.c | 14
main/bucket.c | 2
main/cdr.c | 8
main/channel.c | 40
main/cli.c | 4
main/codec.c | 4
main/codec_builtin.c | 2
main/config.c | 17
main/dial.c | 8
main/endpoints.c | 8
main/format_cap.c | 4
main/http.c | 9
main/json.c | 29
main/loader.c | 79 +
main/logger.c | 31
main/manager.c | 7
main/netsock2.c | 4
main/rtp_engine.c | 2
main/sched.c | 20
main/sdp_srtp.c | 109 +-
main/sorcery.c | 11
main/stasis_bridges.c | 3
main/stasis_channels.c | 6
main/stdtime/localtime.c | 279 ++++-
main/tcptls.c | 5
main/translate.c | 4
main/utils.c | 4
main/xmldoc.c | 9
makeopts.in | 2
res/ari/ari_websockets.c | 14
res/ari/resource_channels.c | 136 ++
res/ari/resource_channels.h | 26
res/ari/resource_endpoints.c | 61 -
res/res_ari_channels.c | 115 ++
res/res_ari_endpoints.c | 1
res/res_config_odbc.c | 8
res/res_format_attr_h264.c | 4
res/res_http_websocket.c | 43
res/res_odbc.c | 11
res/res_pjsip.c | 150 ++-
res/res_pjsip/config_domain_aliases.c | 1
res/res_pjsip/config_global.c | 152 ++-
res/res_pjsip/include/res_pjsip_private.h | 164 +++
res/res_pjsip/pjsip_cli.c | 25
res/res_pjsip/pjsip_configuration.c | 14
res/res_pjsip/pjsip_global_headers.c | 14
res/res_pjsip/pjsip_options.c | 23
res/res_pjsip_acl.c | 3
res/res_pjsip_caller_id.c | 53 -
res/res_pjsip_config_wizard.c | 224 ++++
res/res_pjsip_endpoint_identifier_anonymous.c | 2
res/res_pjsip_endpoint_identifier_ip.c | 2
res/res_pjsip_endpoint_identifier_user.c | 2
res/res_pjsip_exten_state.c | 3
res/res_pjsip_keepalive.c | 2
res/res_pjsip_messaging.c | 24
res/res_pjsip_multihomed.c | 5
res/res_pjsip_nat.c | 12
res/res_pjsip_publish_asterisk.c | 1
res/res_pjsip_pubsub.c | 2
res/res_pjsip_refer.c | 263 +++--
res/res_pjsip_registrar.c | 8
res/res_pjsip_sdp_rtp.c | 22
res/res_pjsip_send_to_voicemail.c | 10
res/res_pjsip_session.c | 488 +++++++---
res/res_pjsip_session.exports.in | 1
res/res_pjsip_t38.c | 4
res/res_pjsip_transport_websocket.c | 2
res/res_rtp_asterisk.c | 20
res/res_sorcery_config.c | 46
res/res_timing_pthread.c | 3
res/stasis/control.c | 32
rest-api/api-docs/channels.json | 53 +
rest-api/api-docs/endpoints.json | 4
tests/test_func_file.c | 6
137 files changed, 4102 insertions(+), 1158 deletions(-)
----------------------------------------------------------------------