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Importing release summary for 13.0.0-beta2 release.

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<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - asterisk-13.0.0-beta2</title></head>
<body>
<h1 align="center"><a name="top">Release Summary</a></h1>
<h3 align="center">asterisk-13.0.0-beta2</h3>
<h3 align="center">Date: 2014-09-19</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 new features. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.</p>
<p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.0.0-beta1.</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>
21 rmudgett<br/>
20 mmichelson<br/>
16 mjordan<br/>
13 jrose<br/>
10 file<br/>
10 gtjoseph<br/>
7 kmoore<br/>
4 jcolp<br/>
3 wdoekes<br/>
2 Jeremy Laine<br/>
2 seanbright<br/>
2 sgriepentrog<br/>
2 sruffell<br/>
1 cloos<br/>
1 dlee<br/>
1 Elazar Broad<br/>
1 elguero<br/>
1 newtonr<br/>
1 wedhorn<br/>
</td>
<td>
2 George Joseph<br/>
1 Damien Wedhorn<br/>
1 David Herselman<br/>
1 Deepak Singh Rawat<br/>
1 dimitripietro<br/>
1 elguero<br/>
1 Kilburn<br/>
1 Samuel Galarneau<br/>
1 sruffell<br/>
1 Tony Lewis<br/>
1 wdoekes<br/>
</td>
<td>
7 mjordan<br/>
3 sruffell<br/>
2 mmichelson<br/>
2 sharky<br/>
1 amohod<br/>
1 ateks<br/>
1 bbs2web<br/>
1 dafi<br/>
1 dimitripietro<br/>
1 dsr<br/>
1 Each<br/>
1 ebroad<br/>
1 edvinv<br/>
1 falves11<br/>
1 jideliov<br/>
1 jrose<br/>
1 krandonbruse<br/>
1 maddog<br/>
1 pnlarsson<br/>
1 proftech<br/>
1 rmudgett<br/>
1 RomanSk<br/>
1 sgalarneau<br/>
1 sgriepentrog<br/>
1 slavon<br/>
1 wdoekes<br/>
1 xrobau<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: . I did not set the category correctly.</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24147">ASTERISK-24147</a>: ARI: channel hangup crashes asterisk process<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421880">421880</a><br/>
Reporter: edvinv<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Applications/app_controlplayback</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421696">421696</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-24225">ASTERISK-24225</a>: Dial option z is broken<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421235">421235</a><br/>
Reporter: dimitripietro<br/>
Testers: dimitripietro<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Applications/app_meetme</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24234">ASTERISK-24234</a>: app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421273">421273</a><br/>
Reporter: sruffell<br/>
Testers: sruffell<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Applications/app_mixmonitor</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420940">420940</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421187">421187</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<h3>Category: CDR/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24237">ASTERISK-24237</a>: CDR: FRACK With PJSIP blonde transfer.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423530">423530</a><br/>
Reporter: rmudgett<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24241">ASTERISK-24241</a>: crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422716">422716</a><br/>
Reporter: dsr<br/>
Testers: Deepak Singh Rawat<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24254">ASTERISK-24254</a>: CDRs: Application/args/dialplan CEP updated during dial operation<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422719">422719</a><br/>
Reporter: mjordan<br/>
Testers: Tony Lewis<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Channels/chan_iax2</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23767">ASTERISK-23767</a>: [patch] Dynamic IAX2 registration stops trying if ever not able to resolve<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422276">422276</a><br/>
Reporter: bbs2web<br/>
Testers: David Herselman, elguero<br/>
Coders: elguero<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24265">ASTERISK-24265</a>: segfault in asterisk when try to make call to IAX <br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423524">423524</a><br/>
Reporter: dafi<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Channels/chan_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24143">ASTERISK-24143</a>: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421956">421956</a><br/>
Reporter: Each<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422542">422542</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Channels/chan_sip/General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24178">ASTERISK-24178</a>: [patch]fromdomainport used even if not set<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421720">421720</a><br/>
Reporter: ebroad<br/>
Coders: Elazar Broad<br/>
<br/>
<h3>Category: Channels/chan_sip/Messaging</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24301">ASTERISK-24301</a>: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423372">423372</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Channels/chan_sip/WebSocket</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23997">ASTERISK-23997</a>: chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421911">421911</a><br/>
Reporter: slavon<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Core/Configuration</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24231">ASTERISK-24231</a>: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422985">422985</a><br/>
Reporter: pnlarsson<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Core/ManagerInterface</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24331">ASTERISK-24331</a>: Unexpected Errors in Asterisk Manager Interface Output<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423284">423284</a><br/>
Reporter: xrobau<br/>
Testers: George Joseph<br/>
Coders: gtjoseph<br/>
<br/>
<h3>Category: Core/PBX</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24249">ASTERISK-24249</a>: SIP debugs do not stop<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423069">423069</a><br/>
Reporter: amohod<br/>
Coders: wdoekes<br/>
<br/>
<h3>Category: Documentation</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422374">422374</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422379">422379</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine<br/>
<br/>
<h3>Category: General</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24032">ASTERISK-24032</a>: Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421230">421230</a><br/>
Reporter: maddog<br/>
Testers: Kilburn, wdoekes<br/>
Coders: cloos<br/>
<br/>
<h3>Category: Resources/res_agi</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420940">420940</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24027">ASTERISK-24027</a>: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421187">421187</a><br/>
Reporter: mjordan<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Resources/res_ari</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24043">ASTERISK-24043</a>: ARI /continue fails to actually continue into the dialplan<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421423">421423</a><br/>
Reporter: krandonbruse<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421696">421696</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24264">ASTERISK-24264</a>: ARI: Adding a channel to a holding bridge automatically starts MOH<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422504">422504</a><br/>
Reporter: sgalarneau<br/>
Testers: Samuel Galarneau<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_ari_bridges</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24264">ASTERISK-24264</a>: ARI: Adding a channel to a holding bridge automatically starts MOH<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422504">422504</a><br/>
Reporter: sgalarneau<br/>
Testers: Samuel Galarneau<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_ari_playbacks</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24229">ASTERISK-24229</a>: ARI: playback of sounds implicitly answers channel, preventing early media playback<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421696">421696</a><br/>
Reporter: mjordan<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_fax</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24301">ASTERISK-24301</a>: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423372">423372</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Resources/res_hep_rtcp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24236">ASTERISK-24236</a>: res_hep_rtcp: Module incorrectly depends on pjsip<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421065">421065</a><br/>
Reporter: mjordan<br/>
Testers: Damien Wedhorn<br/>
Coders: mjordan<br/>
<br/>
<h3>Category: Resources/res_musiconhold</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22252">ASTERISK-22252</a>: res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421788">421788</a><br/>
Reporter: wdoekes<br/>
Coders: jrose<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24019">ASTERISK-24019</a>: When a Music On Hold stream starts it restarts at beginning of file.<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421979">421979</a><br/>
Reporter: ateks<br/>
Coders: rmudgett<br/>
<br/>
<h3>Category: Resources/res_pjsip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24161">ASTERISK-24161</a>: PJSIPShowEndpoint gives inaccurate count of list items<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423284">423284</a><br/>
Reporter: mmichelson<br/>
Testers: George Joseph<br/>
Coders: gtjoseph<br/>
<br/>
<h3>Category: Resources/res_pjsip_endpoint_identifier_ip</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24290">ASTERISK-24290</a>: Endpoint identifier match value fails to parse when CIDR network format is specified<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423425">423425</a><br/>
Reporter: proftech<br/>
Coders: jrose<br/>
<br/>
<h3>Category: Resources/res_pjsip_nat</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23634">ASTERISK-23634</a>: With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423152">423152</a><br/>
Reporter: RomanSk<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_pjsip_pubsub</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24136">ASTERISK-24136</a>: Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423348">423348</a><br/>
Reporter: mmichelson<br/>
Coders: mmichelson<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24181">ASTERISK-24181</a>: RLS: Large lists don't get sent because they exceed the PJSIP message length limit<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422851">422851</a><br/>
Reporter: jrose<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Resources/res_pjsip_sdp_rtp</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23994">ASTERISK-23994</a>: res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421797">421797</a><br/>
Reporter: falves11<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Resources/res_pjsip_transport_websocket</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24143">ASTERISK-24143</a>: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421956">421956</a><br/>
Reporter: Each<br/>
Coders: jcolp<br/>
<br/>
<h3>Category: Resources/res_rtp_asterisk</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23577">ASTERISK-23577</a>: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423152">423152</a><br/>
Reporter: jideliov<br/>
Coders: jcolp<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422542">422542</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Tests/testsuite</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24212">ASTERISK-24212</a>: testsuite: Sporadic crash due to assert on stopping RTP engine<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422542">422542</a><br/>
Reporter: mjordan<br/>
Coders: mmichelson<br/>
<br/>
<h3>Category: Utilities/aelparse</h3><br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422374">422374</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine<br/>
<br/>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24171">ASTERISK-24171</a>: [patch] Provide a manpage for the aelparse utility<br/>
Revision: <a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422379">422379</a><br/>
Reporter: sharky<br/>
Coders: Jeremy Laine<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=420837">420837</a></td><td>rmudgett</td><td>res/stasis/command.c: Fix recent commit using spaces instead of tabs.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420856">420856</a></td><td>file</td><td>app_voicemail: Fix the "test_voicemail_vm_info" unit test.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420879">420879</a></td><td>rmudgett</td><td>res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420881">420881</a></td><td>rmudgett</td><td>chan_sip: Fix type mismatch when the format is changed.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420899">420899</a></td><td>wdoekes</td><td>logger: Don't store verbose-magic in the log files.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420919">420919</a></td><td>kmoore</td><td>AMI: Improve documentation for Status action</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420950">420950</a></td><td>kmoore</td><td>PJSIP: Prevent crash no-URI contacts</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420957">420957</a></td><td>rmudgett</td><td>res_pjsip_send_to_voicemail.c: Fix svn file properties.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=420992">420992</a></td><td>rmudgett</td><td>channel_internal_api.c: Replace some code with ao2_replace().</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421010">421010</a></td><td>rmudgett</td><td>ARI: Originate to app local channel subscription code optimization.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421042">421042</a></td><td>mjordan</td><td>cel: Make sure channels in extra fields include their unique IDs as well</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421062">421062</a></td><td>mjordan</td><td>main/file: Move test event to emit PLAYBACK event more consistently</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421166">421166</a></td><td>mjordan</td><td>app_voicemail/app: Remove test events that were duplicated by r421059</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421210">421210</a></td><td>file</td><td>res_http_websocket: Include query parameters in client connection requests.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421311">421311</a></td><td>mjordan</td><td>res/ari/resource_channels: Don't return allocation failure on failed function</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421312">421312</a></td><td>mjordan</td><td>res/ari/resource_channels: Fix compilation issue</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421337">421337</a></td><td>gtjoseph</td><td>func_config: Change 'Not Found' message from ERROR to DEBUG</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421376">421376</a></td><td>wedhorn</td><td>Skinny: Fixup compile warning for non dev-mode.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421403">421403</a></td><td>rmudgett</td><td>chan_pjsip: Fix attended transfer connected line name update.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421445">421445</a></td><td>kmoore</td><td>AMI Docs: Fix Status channel parameter optionality</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421448">421448</a></td><td>mmichelson</td><td>Fix compilation error on certain versions of GCC.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421488">421488</a></td><td>mmichelson</td><td>Alter documentation for callerid_privacy to use correct values.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421538">421538</a></td><td>kmoore</td><td>Stasis: Add information to blind transfer event</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421566">421566</a></td><td>mmichelson</td><td>Move evaluation of set_var options in pjsip to the end of channel initialization.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421585">421585</a></td><td>mmichelson</td><td>Set the role for inbound subscriptions correctly.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421616">421616</a></td><td>rmudgett</td><td>cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421645">421645</a></td><td>rmudgett</td><td>chan_pjsip: Update media translation paths when new SDP negotiated.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421783">421783</a></td><td>mmichelson</td><td>Improve consistency of party ID privacy usage.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421790">421790</a></td><td>mmichelson</td><td>Let's try checking the name and number, instead of the name twice.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421794">421794</a></td><td>mmichelson</td><td>Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421802">421802</a></td><td>rmudgett</td><td>res_musiconhold.c: Remove obsolete REF_DEBUG code.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421860">421860</a></td><td>mjordan</td><td>main/message: Add a new-line to a DEBUG message</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421882">421882</a></td><td>mmichelson</td><td>Fix a locking inversion in MixMonitor.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421932">421932</a></td><td>file</td><td>res_pjsip_transport_websocket: Ensure secure Websocket clients can be called.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=421945">421945</a></td><td>file</td><td>res_pjsip_transport_websocket: Fix a progressive memory growth.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422037">422037</a></td><td>rmudgett</td><td>res_musiconhold.c: Release any format refs before memset().</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422071">422071</a></td><td>mmichelson</td><td>Fix race condition in the scheduler when deleting a running entry.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422091">422091</a></td><td>gtjoseph</td><td>confbridge: Make kick, mute and unmute handle channel targets consistently.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422154">422154</a></td><td>kmoore</td><td>CallerID: Fix parsing of malformed callerid</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422177">422177</a></td><td>gtjoseph</td><td>confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422200">422200</a></td><td>rmudgett</td><td>sched: Fix typo and whitespace change.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422215">422215</a></td><td>rmudgett</td><td>res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422239">422239</a></td><td>mmichelson</td><td>Fix bug that did not allow for multiple batched RLS notifications to be sent.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422256">422256</a></td><td>rmudgett</td><td>Added ConfBridge AMI event note to UPGRADE.txt.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422296">422296</a></td><td>mjordan</td><td>LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422359">422359</a></td><td>sgriepentrog</td><td>The assertion that peer was not found on final event</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422442">422442</a></td><td>gtjoseph</td><td>manager: Make WaitEvent action respect eventfilters</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422445">422445</a></td><td>gtjoseph</td><td>confbridge: Add Duration to ConfbridgeList event</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422507">422507</a></td><td>mjordan</td><td>main/cli: Do not attempt to show CDR data for internal channels</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422558">422558</a></td><td>file</td><td>res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422631">422631</a></td><td>jrose</td><td>Manager: Require read permission for SYSTEM in order to send FullyBooted</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422646">422646</a></td><td>kmoore</td><td>Menuselect: Fix incorrect enabling on failed deps</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422661">422661</a></td><td>rmudgett</td><td>devicestate.c: Minor tweaks</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422665">422665</a></td><td>jrose</td><td>Call IDs: Fix appearance of call ID in core show channels when NULL</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422684">422684</a></td><td>jrose</td><td>Dial API: Add a dial option to indicate the dialed channel will replace dialer</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422700">422700</a></td><td>rmudgett</td><td>func_channel.c: Add missing locking to some CHANNEL() requests.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422747">422747</a></td><td>file</td><td>res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422767">422767</a></td><td>mjordan</td><td>main/rtp_engine: Format NTP timestamps as unsigned ints</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422770">422770</a></td><td>mjordan</td><td>main/cdr: Copy over location information during a fork</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422836">422836</a></td><td>jrose</td><td>res_pjsip_pubsub: Check supported headers for eventlist when subscribing to</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422853">422853</a></td><td>mmichelson</td><td>Add sample configuration for resource lists.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422855">422855</a></td><td>mmichelson</td><td>Add note about configuring list_items on a single line.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422883">422883</a></td><td>newtonr</td><td>Sounds/BuildSystem: Modifications to include new releases and Japanese language.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422901">422901</a></td><td>seanbright</td><td>pjsip/config_auth.c: Add missing whitespace to log messages.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422905">422905</a></td><td>gtjoseph</td><td>config: bug: fix truncation of included config files on permissions error</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=422965">422965</a></td><td>mmichelson</td><td>Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423129">423129</a></td><td>wdoekes</td><td>contrib: Fix verifyi typo in alembic DB script ps_transport table.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423173">423173</a></td><td>file</td><td>res_pjsip_session: Fix usage of wrong memory pool when creating local SDP.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423209">423209</a></td><td>file</td><td>res_rtp_asterisk: Fix building when pjproject is not used.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423212">423212</a></td><td>file</td><td>res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423255">423255</a></td><td>file</td><td>res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423279">423279</a></td><td>gtjoseph</td><td>config: bug: Fix SEGV in ast_category_insert when matching category isn't found</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423281">423281</a></td><td>dlee</td><td>Only install dahdi_span_config_hook if DAHDI is enabled</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423414">423414</a></td><td>mmichelson</td><td>Add API call to determine if format capability structure is "empty".</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423418">423418</a></td><td>rmudgett</td><td>astobj2.c/refcounter.py: Fix to deal with invalid object refs.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423423">423423</a></td><td>rmudgett</td><td>bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423462">423462</a></td><td>mmichelson</td><td>Add subscription state test events.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423478">423478</a></td><td>gtjoseph</td><td>utils: Create ast_strsep function that ignores separators inside quotes</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423482">423482</a></td><td>seanbright</td><td>res_pjsip: Don't require a password when doing userpass authentication.</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423504">423504</a></td><td>kmoore</td><td>PJSIP: Prevent T38 framehook being put on wrong channel</td>
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/branches/13?view=revision&revision=423561">423561</a></td><td>rmudgett</td><td>res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.</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>
LICENSE | 2
Makefile | 10
UPGRADE.txt | 19
apps/app_chanspy.c | 2
apps/app_confbridge.c | 265 ++-
apps/app_dial.c | 2
apps/app_macro.c | 7
apps/app_meetme.c | 8
apps/app_mixmonitor.c | 7
apps/app_stack.c | 35
apps/app_voicemail.c | 21
apps/confbridge/confbridge_manager.c | 81 +
bridges/bridge_softmix.c | 13
channels/chan_iax2.c | 45
channels/chan_pjsip.c | 102 -
channels/chan_sip.c | 33
channels/chan_skinny.c | 10
configs/samples/pjsip.conf.sample | 50
configs/samples/sip.conf.sample | 4
configure.ac | 4
contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py | 29
contrib/scripts/refcounter.py | 80 -
doc/aelparse.8 | 28
doc/smsq.8 | 146 ++
funcs/func_channel.c | 28
funcs/func_config.c | 2
include/asterisk/channel.h | 43
include/asterisk/config.h | 5
include/asterisk/dial.h | 1
include/asterisk/file.h | 7
include/asterisk/format_cap.h | 9
include/asterisk/framehook.h | 6
include/asterisk/res_pjsip.h | 2
include/asterisk/res_pjsip_pubsub.h | 23
include/asterisk/res_pjsip_session.h | 35
include/asterisk/rtp_engine.h | 4
include/asterisk/sched.h | 2
include/asterisk/stasis_app_impl.h | 25
include/asterisk/stasis_bridges.h | 10
include/asterisk/strings.h | 60
include/asterisk/uri.h | 2
include/asterisk/utils.h | 9
main/app.c | 7
main/astobj2.c | 10
main/bridge.c | 63
main/bridge_after.c | 4
main/bridge_channel.c | 15
main/callerid.c | 63
main/cdr.c | 22
main/cel.c | 27
main/channel.c | 159 +-
main/channel_internal_api.c | 52
main/cli.c | 6
main/config.c | 132 +
main/devicestate.c | 5
main/dial.c | 7
main/dns.c | 3
main/file.c | 9
main/format_cache.c | 3
main/format_cap.c | 15
main/framehook.c | 19
main/logger.c | 24
main/manager.c | 35
main/message.c | 2
main/pbx.c | 5
main/rtp_engine.c | 4
main/sched.c | 49
main/stasis_bridges.c | 28
main/stasis_channels.c | 219 +++
main/translate.c | 5
main/uri.c | 2
main/utils.c | 83 +
makeopts.in | 1
menuselect/menuselect.c | 2
res/ari/ari_model_validators.c | 9
res/ari/ari_model_validators.h | 1
res/ari/resource_channels.c | 17
res/res_fax_spandsp.c | 19
res/res_hep_rtcp.c | 2
res/res_http_websocket.c | 26
res/res_musiconhold.c | 33
res/res_pjsip.c | 10
res/res_pjsip/config_auth.c | 14
res/res_pjsip/config_transport.c | 18
res/res_pjsip/location.c | 2
res/res_pjsip/pjsip_configuration.c | 6
res/res_pjsip/pjsip_options.c | 173 +-
res/res_pjsip_caller_id.c | 94 -
res/res_pjsip_dialog_info_body_generator.c | 1
res/res_pjsip_diversion.c | 1
res/res_pjsip_endpoint_identifier_ip.c | 62
res/res_pjsip_exten_state.c | 8
res/res_pjsip_mwi.c | 13
res/res_pjsip_mwi_body_generator.c | 1
res/res_pjsip_notify.c | 8
res/res_pjsip_pidf_body_generator.c | 1
res/res_pjsip_pubsub.c | 166 ++
res/res_pjsip_sdp_rtp.c | 31
res/res_pjsip_session.c | 72 -
res/res_pjsip_t38.c | 13
res/res_pjsip_transport_websocket.c | 46
res/res_pjsip_xpidf_body_generator.c | 2
res/res_rtp_asterisk.c | 704 ++++++----
res/res_stasis.c | 30
res/res_stasis_answer.c | 2
res/res_stasis_playback.c | 20
res/res_stasis_recording.c | 20
res/res_stasis_snoop.c | 4
res/stasis/app.c | 31
res/stasis/command.c | 41
res/stasis/command.h | 9
res/stasis/control.c | 81 -
res/stasis/messaging.h | 2
res/stasis/stasis_bridge.c | 28
rest-api/api-docs/events.json | 5
sounds/Makefile | 7
sounds/sounds.xml | 27
tests/test_callerid.c | 165 ++
tests/test_cel.c | 21
tests/test_strings.c | 80 +
tests/test_utils.c | 98 +
121 files changed, 3531 insertions(+), 1049 deletions(-)
</pre><br/>
<hr/>
</body>
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Release Summary
asterisk-13.0.0-beta2
Date: 2014-09-19
<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 new features. For a list of new features that have
been included with this release, please see the CHANGES file inside the
source package. Since this is new major release, users are encouraged to
do extended testing before upgrading to this version in a production
environment.
The data in this summary reflects changes that have been made since the
previous release, asterisk-13.0.0-beta1.
----------------------------------------------------------------------
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
21 rmudgett 2 George Joseph 7 mjordan
20 mmichelson 1 Damien Wedhorn 3 sruffell
16 mjordan 1 David Herselman 2 mmichelson
13 jrose 1 Deepak Singh Rawat 2 sharky
10 file 1 dimitripietro 1 amohod
10 gtjoseph 1 elguero 1 ateks
7 kmoore 1 Kilburn 1 bbs2web
4 jcolp 1 Samuel Galarneau 1 dafi
3 wdoekes 1 sruffell 1 dimitripietro
2 Jeremy Laine 1 Tony Lewis 1 dsr
2 seanbright 1 wdoekes 1 Each
2 sgriepentrog 1 ebroad
2 sruffell 1 edvinv
1 cloos 1 falves11
1 dlee 1 jideliov
1 Elazar Broad 1 jrose
1 elguero 1 krandonbruse
1 newtonr 1 maddog
1 wedhorn 1 pnlarsson
1 proftech
1 rmudgett
1 RomanSk
1 sgalarneau
1 sgriepentrog
1 slavon
1 wdoekes
1 xrobau
----------------------------------------------------------------------
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: . I did not set the category correctly.
ASTERISK-24147: ARI: channel hangup crashes asterisk process
Revision: 421880
Reporter: edvinv
Coders: jrose
Category: Applications/app_controlplayback
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan
Category: Applications/app_dial
ASTERISK-24225: Dial option z is broken
Revision: 421235
Reporter: dimitripietro
Testers: dimitripietro
Coders: rmudgett
Category: Applications/app_meetme
ASTERISK-24234: app_meetme: Crash on conference shutdown due to NULL
channel passed to meetme_stasis_generate_msg()
Revision: 421273
Reporter: sruffell
Testers: sruffell
Coders: mjordan
Category: Applications/app_mixmonitor
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
bridge feature causes channel to leave AGI has hung up
Revision: 420940
Reporter: mjordan
Coders: jrose
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
bridge feature causes channel to leave AGI has hung up
Revision: 421187
Reporter: mjordan
Coders: jrose
Category: CDR/General
ASTERISK-24237: CDR: FRACK With PJSIP blonde transfer.
Revision: 423530
Reporter: rmudgett
Coders: jrose
ASTERISK-24241: crash: CDRs recursively attempt to update Party B
information in a multi-party bridge, overrunning the stack
Revision: 422716
Reporter: dsr
Testers: Deepak Singh Rawat
Coders: mjordan
ASTERISK-24254: CDRs: Application/args/dialplan CEP updated during dial
operation
Revision: 422719
Reporter: mjordan
Testers: Tony Lewis
Coders: mjordan
Category: Channels/chan_iax2
ASTERISK-23767: [patch] Dynamic IAX2 registration stops trying if ever not
able to resolve
Revision: 422276
Reporter: bbs2web
Testers: David Herselman, elguero
Coders: elguero
ASTERISK-24265: segfault in asterisk when try to make call to IAX
Revision: 423524
Reporter: dafi
Coders: jrose
Category: Channels/chan_pjsip
ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on
received 200 OK
Revision: 421956
Reporter: Each
Coders: jcolp
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson
Category: Channels/chan_sip/General
ASTERISK-24178: [patch]fromdomainport used even if not set
Revision: 421720
Reporter: ebroad
Coders: Elazar Broad
Category: Channels/chan_sip/Messaging
ASTERISK-24301: Security: Out of call MESSAGE requests processed via
Message channel driver can crash Asterisk
Revision: 423372
Reporter: mjordan
Coders: mmichelson
Category: Channels/chan_sip/WebSocket
ASTERISK-23997: chan_sip: port incorrectly incremented for RTCP ICE
candidates in SDP answer
Revision: 421911
Reporter: slavon
Coders: jcolp
Category: Core/Configuration
ASTERISK-24231: crash: CLI execution of realtime destroy sippeers id 1
causes crash due to NULL name provided to ast_variable
Revision: 422985
Reporter: pnlarsson
Coders: jrose
Category: Core/ManagerInterface
ASTERISK-24331: Unexpected Errors in Asterisk Manager Interface Output
Revision: 423284
Reporter: xrobau
Testers: George Joseph
Coders: gtjoseph
Category: Core/PBX
ASTERISK-24249: SIP debugs do not stop
Revision: 423069
Reporter: amohod
Coders: wdoekes
Category: Documentation
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422374
Reporter: sharky
Coders: Jeremy Laine
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422379
Reporter: sharky
Coders: Jeremy Laine
Category: General
ASTERISK-24032: Gentoo compilation emits warning: "_FORTIFY_SOURCE"
redefined
Revision: 421230
Reporter: maddog
Testers: Kilburn, wdoekes
Coders: cloos
Category: Resources/res_agi
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
bridge feature causes channel to leave AGI has hung up
Revision: 420940
Reporter: mjordan
Coders: jrose
ASTERISK-24027: MixMonitor AMI action called during AGI execution from
bridge feature causes channel to leave AGI has hung up
Revision: 421187
Reporter: mjordan
Coders: jrose
Category: Resources/res_ari
ASTERISK-24043: ARI /continue fails to actually continue into the dialplan
Revision: 421423
Reporter: krandonbruse
Coders: jrose
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically
starts MOH
Revision: 422504
Reporter: sgalarneau
Testers: Samuel Galarneau
Coders: mjordan
Category: Resources/res_ari_bridges
ASTERISK-24264: ARI: Adding a channel to a holding bridge automatically
starts MOH
Revision: 422504
Reporter: sgalarneau
Testers: Samuel Galarneau
Coders: mjordan
Category: Resources/res_ari_playbacks
ASTERISK-24229: ARI: playback of sounds implicitly answers channel,
preventing early media playback
Revision: 421696
Reporter: mjordan
Coders: mjordan
Category: Resources/res_fax
ASTERISK-24301: Security: Out of call MESSAGE requests processed via
Message channel driver can crash Asterisk
Revision: 423372
Reporter: mjordan
Coders: mmichelson
Category: Resources/res_hep_rtcp
ASTERISK-24236: res_hep_rtcp: Module incorrectly depends on pjsip
Revision: 421065
Reporter: mjordan
Testers: Damien Wedhorn
Coders: mjordan
Category: Resources/res_musiconhold
ASTERISK-22252: res_musiconhold cleanup - REF_DEBUG reload warnings and
ref leaks
Revision: 421788
Reporter: wdoekes
Coders: jrose
ASTERISK-24019: When a Music On Hold stream starts it restarts at
beginning of file.
Revision: 421979
Reporter: ateks
Coders: rmudgett
Category: Resources/res_pjsip
ASTERISK-24161: PJSIPShowEndpoint gives inaccurate count of list items
Revision: 423284
Reporter: mmichelson
Testers: George Joseph
Coders: gtjoseph
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-24290: Endpoint identifier match value fails to parse when CIDR
network format is specified
Revision: 423425
Reporter: proftech
Coders: jrose
Category: Resources/res_pjsip_nat
ASTERISK-23634: With TURN Asterisk crashes on multiple (7-10) concurrent
WebRTC (avpg/encryption/icesupport) calls
Revision: 423152
Reporter: RomanSk
Coders: jcolp
Category: Resources/res_pjsip_pubsub
ASTERISK-24136: Security: Crash in Asterisk's PJSIP code when subscribing
to an event with an unexpected body type
Revision: 423348
Reporter: mmichelson
Coders: mmichelson
ASTERISK-24181: RLS: Large lists don't get sent because they exceed the
PJSIP message length limit
Revision: 422851
Reporter: jrose
Coders: mmichelson
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-23994: res_pjsip_sdp_rtp: owner address in SDP may not be fully
qualified domainname
Revision: 421797
Reporter: falves11
Coders: mmichelson
Category: Resources/res_pjsip_transport_websocket
ASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on
received 200 OK
Revision: 421956
Reporter: Each
Coders: jcolp
Category: Resources/res_rtp_asterisk
ASTERISK-23577: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when
RTP instance is NULL
Revision: 423152
Reporter: jideliov
Coders: jcolp
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson
Category: Tests/testsuite
ASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP
engine
Revision: 422542
Reporter: mjordan
Coders: mmichelson
Category: Utilities/aelparse
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422374
Reporter: sharky
Coders: Jeremy Laine
ASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Revision: 422379
Reporter: sharky
Coders: Jeremy Laine
----------------------------------------------------------------------
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/stasis/command.c: Fix | |
| 420837 | rmudgett | recent commit using spaces | |
| | | instead of tabs. | |
|----------+--------------+---------------------------------+------------|
| | | app_voicemail: Fix the | |
| 420856 | file | "test_voicemail_vm_info" unit | |
| | | test. | |
|----------+--------------+---------------------------------+------------|
| | | res_stasis_snoop.c: Fix off | |
| 420879 | rmudgett | nominial exit path leaving | |
| | | Snoop channel locked and not | |
| | | hungup. | |
|----------+--------------+---------------------------------+------------|
| 420881 | rmudgett | chan_sip: Fix type mismatch | |
| | | when the format is changed. | |
|----------+--------------+---------------------------------+------------|
| 420899 | wdoekes | logger: Don't store | |
| | | verbose-magic in the log files. | |
|----------+--------------+---------------------------------+------------|
| 420919 | kmoore | AMI: Improve documentation for | |
| | | Status action | |
|----------+--------------+---------------------------------+------------|
| 420950 | kmoore | PJSIP: Prevent crash no-URI | |
| | | contacts | |
|----------+--------------+---------------------------------+------------|
| 420957 | rmudgett | res_pjsip_send_to_voicemail.c: | |
| | | Fix svn file properties. | |
|----------+--------------+---------------------------------+------------|
| 420992 | rmudgett | channel_internal_api.c: Replace | |
| | | some code with ao2_replace(). | |
|----------+--------------+---------------------------------+------------|
| | | ARI: Originate to app local | |
| 421010 | rmudgett | channel subscription code | |
| | | optimization. | |
|----------+--------------+---------------------------------+------------|
| | | cel: Make sure channels in | |
| 421042 | mjordan | extra fields include their | |
| | | unique IDs as well | |
|----------+--------------+---------------------------------+------------|
| | | main/file: Move test event to | |
| 421062 | mjordan | emit PLAYBACK event more | |
| | | consistently | |
|----------+--------------+---------------------------------+------------|
| | | app_voicemail/app: Remove test | |
| 421166 | mjordan | events that were duplicated by | |
| | | r421059 | |
|----------+--------------+---------------------------------+------------|
| | | res_http_websocket: Include | |
| 421210 | file | query parameters in client | |
| | | connection requests. | |
|----------+--------------+---------------------------------+------------|
| | | res/ari/resource_channels: | |
| 421311 | mjordan | Don't return allocation failure | |
| | | on failed function | |
|----------+--------------+---------------------------------+------------|
| 421312 | mjordan | res/ari/resource_channels: Fix | |
| | | compilation issue | |
|----------+--------------+---------------------------------+------------|
| 421337 | gtjoseph | func_config: Change 'Not Found' | |
| | | message from ERROR to DEBUG | |
|----------+--------------+---------------------------------+------------|
| 421376 | wedhorn | Skinny: Fixup compile warning | |
| | | for non dev-mode. | |
|----------+--------------+---------------------------------+------------|
| | | chan_pjsip: Fix attended | |
| 421403 | rmudgett | transfer connected line name | |
| | | update. | |
|----------+--------------+---------------------------------+------------|
| 421445 | kmoore | AMI Docs: Fix Status channel | |
| | | parameter optionality | |
|----------+--------------+---------------------------------+------------|
| 421448 | mmichelson | Fix compilation error on | |
| | | certain versions of GCC. | |
|----------+--------------+---------------------------------+------------|
| | | Alter documentation for | |
| 421488 | mmichelson | callerid_privacy to use correct | |
| | | values. | |
|----------+--------------+---------------------------------+------------|
| 421538 | kmoore | Stasis: Add information to | |
| | | blind transfer event | |
|----------+--------------+---------------------------------+------------|
| | | Move evaluation of set_var | |
| 421566 | mmichelson | options in pjsip to the end of | |
| | | channel initialization. | |
|----------+--------------+---------------------------------+------------|
| 421585 | mmichelson | Set the role for inbound | |
| | | subscriptions correctly. | |
|----------+--------------+---------------------------------+------------|
| | | cli.c: Fix tab completion of | |
| 421616 | rmudgett | "module load" when MALLOC_DEBUG | |
| | | is enabled. | |
|----------+--------------+---------------------------------+------------|
| | | chan_pjsip: Update media | |
| 421645 | rmudgett | translation paths when new SDP | |
| | | negotiated. | |
|----------+--------------+---------------------------------+------------|
| 421783 | mmichelson | Improve consistency of party ID | |
| | | privacy usage. | |
|----------+--------------+---------------------------------+------------|
| | | Let's try checking the name and | |
| 421790 | mmichelson | number, instead of the name | |
| | | twice. | |
|----------+--------------+---------------------------------+------------|
| | | Ensure after-bridge behavior is | |
| 421794 | mmichelson | correct when moving from Stasis | |
| | | to a non-Stasis bridge. | |
|----------+--------------+---------------------------------+------------|
| 421802 | rmudgett | res_musiconhold.c: Remove | |
| | | obsolete REF_DEBUG code. | |
|----------+--------------+---------------------------------+------------|
| 421860 | mjordan | main/message: Add a new-line to | |
| | | a DEBUG message | |
|----------+--------------+---------------------------------+------------|
| 421882 | mmichelson | Fix a locking inversion in | |
| | | MixMonitor. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_transport_websocket: | |
| 421932 | file | Ensure secure Websocket clients | |
| | | can be called. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_transport_websocket: | |
| 421945 | file | Fix a progressive memory | |
| | | growth. | |
|----------+--------------+---------------------------------+------------|
| 422037 | rmudgett | res_musiconhold.c: Release any | |
| | | format refs before memset(). | |
|----------+--------------+---------------------------------+------------|
| | | Fix race condition in the | |
| 422071 | mmichelson | scheduler when deleting a | |
| | | running entry. | |
|----------+--------------+---------------------------------+------------|
| | | confbridge: Make kick, mute and | |
| 422091 | gtjoseph | unmute handle channel targets | |
| | | consistently. | |
|----------+--------------+---------------------------------+------------|
| 422154 | kmoore | CallerID: Fix parsing of | |
| | | malformed callerid | |
|----------+--------------+---------------------------------+------------|
| | | confbridge: Add 'Admin' param | |
| 422177 | gtjoseph | to join, leave, mute, unmute | |
| | | and talking events | |
|----------+--------------+---------------------------------+------------|
| 422200 | rmudgett | sched: Fix typo and whitespace | |
| | | change. | |
|----------+--------------+---------------------------------+------------|
| | | res/res_pjsip/pjsip_options.c: | |
| 422215 | rmudgett | Eliminate excessive RAII_VAR | |
| | | usage. | |
|----------+--------------+---------------------------------+------------|
| | | Fix bug that did not allow for | |
| 422239 | mmichelson | multiple batched RLS | |
| | | notifications to be sent. | |
|----------+--------------+---------------------------------+------------|
| 422256 | rmudgett | Added ConfBridge AMI event note | |
| | | to UPGRADE.txt. | |
|----------+--------------+---------------------------------+------------|
| | | LICENSE: Clarify language in | |
| 422296 | mjordan | Asterisk's LICENSE to allow for | |
| | | linking to UniMRCP | |
|----------+--------------+---------------------------------+------------|
| 422359 | sgriepentrog | The assertion that peer was not | |
| | | found on final event | |
|----------+--------------+---------------------------------+------------|
| 422442 | gtjoseph | manager: Make WaitEvent action | |
| | | respect eventfilters | |
|----------+--------------+---------------------------------+------------|
| 422445 | gtjoseph | confbridge: Add Duration to | |
| | | ConfbridgeList event | |
|----------+--------------+---------------------------------+------------|
| | | main/cli: Do not attempt to | |
| 422507 | mjordan | show CDR data for internal | |
| | | channels | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_transport_websocket: | |
| 422558 | file | Fix crash when the Contact | |
| | | header is not a URI. | |
|----------+--------------+---------------------------------+------------|
| | | Manager: Require read | |
| 422631 | jrose | permission for SYSTEM in order | |
| | | to send FullyBooted | |
|----------+--------------+---------------------------------+------------|
| 422646 | kmoore | Menuselect: Fix incorrect | |
| | | enabling on failed deps | |
|----------+--------------+---------------------------------+------------|
| 422661 | rmudgett | devicestate.c: Minor tweaks | |
|----------+--------------+---------------------------------+------------|
| | | Call IDs: Fix appearance of | |
| 422665 | jrose | call ID in core show channels | |
| | | when NULL | |
|----------+--------------+---------------------------------+------------|
| | | Dial API: Add a dial option to | |
| 422684 | jrose | indicate the dialed channel | |
| | | will replace dialer | |
|----------+--------------+---------------------------------+------------|
| | | func_channel.c: Add missing | |
| 422700 | rmudgett | locking to some CHANNEL() | |
| | | requests. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_sdp_rtp: Fix | |
| 422747 | file | retrieval of "ice-pwd" | |
| | | attribute if in session and not | |
| | | media stream. | |
|----------+--------------+---------------------------------+------------|
| 422767 | mjordan | main/rtp_engine: Format NTP | |
| | | timestamps as unsigned ints | |
|----------+--------------+---------------------------------+------------|
| 422770 | mjordan | main/cdr: Copy over location | |
| | | information during a fork | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_pubsub: Check | |
| 422836 | jrose | supported headers for eventlist | |
| | | when subscribing to | |
|----------+--------------+---------------------------------+------------|
| 422853 | mmichelson | Add sample configuration for | |
| | | resource lists. | |
|----------+--------------+---------------------------------+------------|
| 422855 | mmichelson | Add note about configuring | |
| | | list_items on a single line. | |
|----------+--------------+---------------------------------+------------|
| | | Sounds/BuildSystem: | |
| 422883 | newtonr | Modifications to include new | |
| | | releases and Japanese language. | |
|----------+--------------+---------------------------------+------------|
| | | pjsip/config_auth.c: Add | |
| 422901 | seanbright | missing whitespace to log | |
| | | messages. | |
|----------+--------------+---------------------------------+------------|
| | | config: bug: fix truncation of | |
| 422905 | gtjoseph | included config files on | |
| | | permissions error | |
|----------+--------------+---------------------------------+------------|
| | | Remove undocumented default | |
| 422965 | mmichelson | behavior of | |
| | | ast_play_and_record_full | |
| | | acceptdtmf. | |
|----------+--------------+---------------------------------+------------|
| | | contrib: Fix verifyi typo in | |
| 423129 | wdoekes | alembic DB script ps_transport | |
| | | table. | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_session: Fix usage of | |
| 423173 | file | wrong memory pool when creating | |
| | | local SDP. | |
|----------+--------------+---------------------------------+------------|
| 423209 | file | res_rtp_asterisk: Fix building | |
| | | when pjproject is not used. | |
|----------+--------------+---------------------------------+------------|
| | | res_rtp_asterisk: Fix 100% CPU | |
| 423212 | file | usage due to timer heap thread | |
| | | spinning. | |
|----------+--------------+---------------------------------+------------|
| | | res_rtp_asterisk: Ensure that | |
| 423255 | file | the thread terminating pj stuff | |
| | | is registered. | |
|----------+--------------+---------------------------------+------------|
| | | config: bug: Fix SEGV in | |
| 423279 | gtjoseph | ast_category_insert when | |
| | | matching category isn't found | |
|----------+--------------+---------------------------------+------------|
| | | Only install | |
| 423281 | dlee | dahdi_span_config_hook if DAHDI | |
| | | is enabled | |
|----------+--------------+---------------------------------+------------|
| | | Add API call to determine if | |
| 423414 | mmichelson | format capability structure is | |
| | | "empty". | |
|----------+--------------+---------------------------------+------------|
| 423418 | rmudgett | astobj2.c/refcounter.py: Fix to | |
| | | deal with invalid object refs. | |
|----------+--------------+---------------------------------+------------|
| | | bridge_softmix.c: Made use | |
| 423423 | rmudgett | ao2_replace() instead of the | |
| | | inline equivalent. | |
|----------+--------------+---------------------------------+------------|
| 423462 | mmichelson | Add subscription state test | |
| | | events. | |
|----------+--------------+---------------------------------+------------|
| | | utils: Create ast_strsep | |
| 423478 | gtjoseph | function that ignores | |
| | | separators inside quotes | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip: Don't require a | |
| 423482 | seanbright | password when doing userpass | |
| | | authentication. | |
|----------+--------------+---------------------------------+------------|
| 423504 | kmoore | PJSIP: Prevent T38 framehook | |
| | | being put on wrong channel | |
|----------+--------------+---------------------------------+------------|
| | | res_pjsip_sdp_rtp.c: Fix native | |
| 423561 | rmudgett | formats containing formats that | |
| | | were not negotiated. | |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
LICENSE | 2
Makefile | 10
UPGRADE.txt | 19
apps/app_chanspy.c | 2
apps/app_confbridge.c | 265 ++-
apps/app_dial.c | 2
apps/app_macro.c | 7
apps/app_meetme.c | 8
apps/app_mixmonitor.c | 7
apps/app_stack.c | 35
apps/app_voicemail.c | 21
apps/confbridge/confbridge_manager.c | 81 +
bridges/bridge_softmix.c | 13
channels/chan_iax2.c | 45
channels/chan_pjsip.c | 102 -
channels/chan_sip.c | 33
channels/chan_skinny.c | 10
configs/samples/pjsip.conf.sample | 50
configs/samples/sip.conf.sample | 4
configure.ac | 4
contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py | 29
contrib/scripts/refcounter.py | 80 -
doc/aelparse.8 | 28
doc/smsq.8 | 146 ++
funcs/func_channel.c | 28
funcs/func_config.c | 2
include/asterisk/channel.h | 43
include/asterisk/config.h | 5
include/asterisk/dial.h | 1
include/asterisk/file.h | 7
include/asterisk/format_cap.h | 9
include/asterisk/framehook.h | 6
include/asterisk/res_pjsip.h | 2
include/asterisk/res_pjsip_pubsub.h | 23
include/asterisk/res_pjsip_session.h | 35
include/asterisk/rtp_engine.h | 4
include/asterisk/sched.h | 2
include/asterisk/stasis_app_impl.h | 25
include/asterisk/stasis_bridges.h | 10
include/asterisk/strings.h | 60
include/asterisk/uri.h | 2
include/asterisk/utils.h | 9
main/app.c | 7
main/astobj2.c | 10
main/bridge.c | 63
main/bridge_after.c | 4
main/bridge_channel.c | 15
main/callerid.c | 63
main/cdr.c | 22
main/cel.c | 27
main/channel.c | 159 +-
main/channel_internal_api.c | 52
main/cli.c | 6
main/config.c | 132 +
main/devicestate.c | 5
main/dial.c | 7
main/dns.c | 3
main/file.c | 9
main/format_cache.c | 3
main/format_cap.c | 15
main/framehook.c | 19
main/logger.c | 24
main/manager.c | 35
main/message.c | 2
main/pbx.c | 5
main/rtp_engine.c | 4
main/sched.c | 49
main/stasis_bridges.c | 28
main/stasis_channels.c | 219 +++
main/translate.c | 5
main/uri.c | 2
main/utils.c | 83 +
makeopts.in | 1
menuselect/menuselect.c | 2
res/ari/ari_model_validators.c | 9
res/ari/ari_model_validators.h | 1
res/ari/resource_channels.c | 17
res/res_fax_spandsp.c | 19
res/res_hep_rtcp.c | 2
res/res_http_websocket.c | 26
res/res_musiconhold.c | 33
res/res_pjsip.c | 10
res/res_pjsip/config_auth.c | 14
res/res_pjsip/config_transport.c | 18
res/res_pjsip/location.c | 2
res/res_pjsip/pjsip_configuration.c | 6
res/res_pjsip/pjsip_options.c | 173 +-
res/res_pjsip_caller_id.c | 94 -
res/res_pjsip_dialog_info_body_generator.c | 1
res/res_pjsip_diversion.c | 1
res/res_pjsip_endpoint_identifier_ip.c | 62
res/res_pjsip_exten_state.c | 8
res/res_pjsip_mwi.c | 13
res/res_pjsip_mwi_body_generator.c | 1
res/res_pjsip_notify.c | 8
res/res_pjsip_pidf_body_generator.c | 1
res/res_pjsip_pubsub.c | 166 ++
res/res_pjsip_sdp_rtp.c | 31
res/res_pjsip_session.c | 72 -
res/res_pjsip_t38.c | 13
res/res_pjsip_transport_websocket.c | 46
res/res_pjsip_xpidf_body_generator.c | 2
res/res_rtp_asterisk.c | 704 ++++++----
res/res_stasis.c | 30
res/res_stasis_answer.c | 2
res/res_stasis_playback.c | 20
res/res_stasis_recording.c | 20
res/res_stasis_snoop.c | 4
res/stasis/app.c | 31
res/stasis/command.c | 41
res/stasis/command.h | 9
res/stasis/control.c | 81 -
res/stasis/messaging.h | 2
res/stasis/stasis_bridge.c | 28
rest-api/api-docs/events.json | 5
sounds/Makefile | 7
sounds/sounds.xml | 27
tests/test_callerid.c | 165 ++
tests/test_cel.c | 21
tests/test_strings.c | 80 +
tests/test_utils.c | 98 +
121 files changed, 3531 insertions(+), 1049 deletions(-)
----------------------------------------------------------------------