-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-13.17.0
-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.17.0-rc1</h3><h3 align="center">Date: 2017-07-06</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
+<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-13.17.0
</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.17.0</h3><h3 align="center">Date: 2017-07-12</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-13.16.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
-<tr valign="top"><td width="33%">17 Sean Bright <sean.bright@gmail.com><br/>1
1 George Joseph <gjoseph@digium.com><br/>10 Joshua Colp <jcolp@digium.com><br/>9 Alexei Gradinari <alex2grad@gmail.com><br/>5 Richard Mudgett <rmudgett@digium.com><br/>5 Kevin Harwell <kharwell@digium.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>1 Jan Friesse <jfriesse@redhat.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/>1 David M. Lee <dlee@digium.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Rodrigo Ramirez Norambuena <a@rodrigoramirez.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Corey Farrell <git@cfware.com><br/></td><td width="33%"><td width="33%">4 Alexei Gradinari <alex2grad@gmail.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Louis Jocelyn Paquet <ljpaquet@quebecinternet.net><br/>3 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>3 George Joseph <gjoseph@digium.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Michael Walton <mike@farsouthnet.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Jacek Konieczny <jkonieczny@eggsoft.pl><br/>1 Tim Morgan <morganuci@gmail.com><br/>1 Etienne Allovon <eallovon@avencall.com><br/>1 alex <asterisk@maximum.guru><br/>1 Kinsey Moore <kmoore@digium.com><br/>1 John Harris <john.harris@certus-tech.com><br/>1 Javier Riveros <goseeped@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Chris Howard <choward@digium.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Andrew Nowrot <andrew.nowrot@gmail.com><br/>1 'alex'<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Ben Langfeld <ben@langfeld.me><br/>1 John Fawcett <john@voipsupport.it><br/>1 Corey Farrell <git@cfware.com><br/>1 Frankie Chin <fchin@biamp.com><br/>1 Zach R <zrothy@monmouth.com><br/>1 Matthias Binder <it@mitterhuemer.at><br/>1 Christopher van de Sande <cvandesande@opendmz.com><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Antoine Pitrou <pitrou@free.fr><br/>1 Alex <metsys@gmx.com><br/>1 Etienne Lessard <elessard97@gmail.com><br/>1 Ryan Smith <ryan.smith@tekara.co.uk><br/>1 Michael Maier <m1278468@mailbox.org><br/>1 OpenBSD ports<br/>1 Marek Cervenka <marek.cervenka@gmail.com><br/>1 Ronald Raikes <reraikes@avweb.com><br/>1 Ove Aursand <oveaurs@gmail.com><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 wushumasters <wushumasters@gmail.com><br/>1 Tony Mountifield <tony@softins.co.uk><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Michel R. Vaillancourt <michel@jkl5group.com><br/>1 David Brillert <david_brillert@scopserv.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/></td></tr>
+<tr valign="top"><td width="33%">17 Sean Bright <sean.bright@gmail.com><br/>1
2 George Joseph <gjoseph@digium.com><br/>10 Joshua Colp <jcolp@digium.com><br/>9 Alexei Gradinari <alex2grad@gmail.com><br/>5 Richard Mudgett <rmudgett@digium.com><br/>5 Kevin Harwell <kharwell@digium.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>1 Jan Friesse <jfriesse@redhat.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/>1 David M. Lee <dlee@digium.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Rodrigo Ramirez Norambuena <a@rodrigoramirez.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Corey Farrell <git@cfware.com><br/></td><td width="33%"><td width="33%">4 Alexei Gradinari <alex2grad@gmail.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Louis Jocelyn Paquet <ljpaquet@quebecinternet.net><br/>3 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>3 George Joseph <gjoseph@digium.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Michael Walton <mike@farsouthnet.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Jacek Konieczny <jkonieczny@eggsoft.pl><br/>1 Tim Morgan <morganuci@gmail.com><br/>1 Etienne Allovon <eallovon@avencall.com><br/>1 alex <asterisk@maximum.guru><br/>1 Kinsey Moore <kmoore@digium.com><br/>1 John Harris <john.harris@certus-tech.com><br/>1 Javier Riveros <goseeped@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Chris Howard <choward@digium.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Andrew Nowrot <andrew.nowrot@gmail.com><br/>1 'alex'<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Ben Langfeld <ben@langfeld.me><br/>1 John Fawcett <john@voipsupport.it><br/>1 Corey Farrell <git@cfware.com><br/>1 Frankie Chin <fchin@biamp.com><br/>1 Zach R <zrothy@monmouth.com><br/>1 Matthias Binder <it@mitterhuemer.at><br/>1 Christopher van de Sande <cvandesande@opendmz.com><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Antoine Pitrou <pitrou@free.fr><br/>1 Alex <metsys@gmx.com><br/>1 Etienne Lessard <elessard97@gmail.com><br/>1 Ryan Smith <ryan.smith@tekara.co.uk><br/>1 Michael Maier <m1278468@mailbox.org><br/>1 OpenBSD ports<br/>1 Marek Cervenka <marek.cervenka@gmail.com><br/>1 Ronald Raikes <reraikes@avweb.com><br/>1 Ove Aursand <oveaurs@gmail.com><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 wushumasters <wushumasters@gmail.com><br/>1 Tony Mountifield <tony@softins.co.uk><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Michel R. Vaillancourt <michel@jkl5group.com><br/>1 David Brillert <david_brillert@scopserv.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Addons/format_mp3</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23951">ASTERISK-23951</a>: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97b003f5e2d4a350508fc20173e180a23f8ef525">[97b003f5e2]</a> Sean Bright -- format_mp3: Re-work menuselect/build issues</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72213c98e3d4d5287ed321f1b4fb67087a7a129c">[72213c98e3]</a> Sean Bright -- format_mp3: Don't try to build format_mp3 if we don't have sources</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25665">ASTERISK-25665</a>: Duplicate logging in queue log for EXITEMPTY events<br/>Reported by: Ove Aursand<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27065">ASTERISK-27065</a>: call hangup after leaving app_queue<br/>Reported by: Marek Cervenka<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26400">ASTERISK-26400</a>: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=996a4791ff123e80d71d44cb0fd13bb201d197b1">[996a4791ff]</a> Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26281">ASTERISK-26281</a>: chan_pjsip would send INVITE to 'Unreachable' endpoints<br/>Reported by: Jacek Konieczny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=746c2c574578608a6b48d4794ba33cda5a6dd484">[746c2c5745]</a> Joshua Colp -- res_pjsip: Add support for returning only reachable contacts and use it.</li>
-</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
+</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27106">ASTERISK-27106</a>: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br/>Reported by: Alexander Traud<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39d2ebbf56635355432eb96ff850c0c9bf2a5d63">[39d2ebbf56]</a> Alexander Traud -- chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).</li>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9f4b3b966e911fae157a484d8f4a1440130eede6">[9f4b3b966e]</a> Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4479038073e57a67c19c1ec5dc8896fcc8c3a0fb">[4479038073]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
</ul><br><h4>Category: Channels/chan_sip/SRTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=67664fbf95a00ced30f8791fd1089b4595e29479">[67664fbf95]</a> Kevin Harwell -- bridge: stuck channel(s) after failed attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
-</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27074">ASTERISK-27074</a>: core_local: local channel data not being properly unref'ed and unlocked<br/>Reported by: Kevin Harwell<ul>
+</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27100">ASTERISK-27100</a>: channel: ast_waitfordigit_full fails to clear flag in an error branch.<br/>Reported by: Corey Farrell<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73520e9f58857049a086fb88106e342cdc25d3a1">[73520e9f58]</a> Corey Farrell -- channel: Clear channel flag in error branch.</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27074">ASTERISK-27074</a>: core_local: local channel data not being properly unref'ed and unlocked<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f9913f2723cbcbf6d78f4da7ee4dd4decc13c05">[1f9913f272]</a> Kevin Harwell -- core_local: local channel data not being properly unref'ed and unlocked</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=161820396495a549c9a378d32136cbb5f28ef2af">[1618203964]</a> Joshua Colp -- asterisk: Audit locking of channel when manipulating flags.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27041">ASTERISK-27041</a>: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc307af7f2ed653914aeadb0b7e613cb4e239b06">[dc307af7f2]</a> Frederic LE FOLL -- Core/PBX: Deadlock between dialplan execution and application unregistration.</li>
-</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
+</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26978">ASTERISK-26978</a>: rtp: Crash in ast_rtp_codecs_payload_code()<br/>Reported by: Ross Beer<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb48e99bd4f4556424a6799e2e5f7aebf8911e8d">[eb48e99bd4]</a> George Joseph -- bridge_native_rtp: Keep rtp instance refs on bridge_channel</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27057">ASTERISK-27057</a>: Seg Fault in ast_sorcery_object_get_id at sorcery.c<br/>Reported by: Ryan Smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2eea791e4178e5f2e4446a5f70d81ac27cf2a0e">[c2eea791e4]</a> George Joseph -- res_pjsip_pubsub: Fix reference to released endpoint</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839">ASTERISK-23839</a>: AGI - RECORD FILE - documentation doesn't describe BEEP argument<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3eb7fbba72482b3019a7493c68e533e67d9d8235">[3eb7fbba72]</a> Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation</li>
-</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27060">ASTERISK-27060</a>: Comment typo format_g729.c<br/>Reported by: Matthew Fredrickson<ul>
+</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27108">ASTERISK-27108</a>: Crash using 'data get' CLI command<br/>Reported by: Sean Bright<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6258de458b2e6ba02e91ed67bbd2801f0984526a">[6258de458b]</a> Sean Bright -- core: Fix segfault when invoking 'data get' CLI command</li>
+</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27060">ASTERISK-27060</a>: Comment typo format_g729.c<br/>Reported by: Matthew Fredrickson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a40073750b46ae28ddf1041d5ed3ab57151298e">[0a40073750]</a> Matthew Fredrickson -- formats/format_g729: Fix typo in comment</li>
</ul><br><h4>Category: PBX/pbx_realtime</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19291">ASTERISK-19291</a>: Background in realtime<br/>Reported by: Andrew Nowrot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=283cc59af746896a2b2bc23899fc86118895f7c0">[283cc59af7]</a> Sean Bright -- pbx_builtin: Properly handle hangup during Background</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=700ef6861ab966008ca16e5f23c64eb68b047c08">[700ef6861a]</a> Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp attributes</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
+</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27090">ASTERISK-27090</a>: PJSIP: Deadlock using TCP transport<br/>Reported by: Richard Mudgett<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d64cbde5756eaa1c7ee62116e112b7ebd198bbe">[0d64cbde57]</a> Richard Mudgett -- pjsip_distributor.c: Fix deadlock with TCP type transports.</li>
</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27052">ASTERISK-27052</a>: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br/>Reported by: alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0bde568669ac26735c1058115ae96223a7e69a6b">[0bde568669]</a> George Joseph -- pjproject_bundled: Use the asterisk github mirror for download</li>
</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053">ASTERISK-27053</a>: res_pjsip_refer/session: Calls dropped during transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26333">ASTERISK-26333</a>: Problems with Blind Transfer, PJSIP (Aastra 6869i)<br/>Reported by: Matthias Binder<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6af2dd34afc2c20bdabd07bc3836821690db4c86">[6af2dd34af]</a> Alexei Gradinari -- res_pjsip: New endpoint option "refer_blind_progress"</li>
-</ul><br><h3>Information Request</h3><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26976">ASTERISK-26976</a>: libsrtp-2.x.x support<br/>Reported by: Alex<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27043">ASTERISK-27043</a>: Core/BuildSystem: Add defines to fix build with LibreSSL<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a64f65fe6fee96702668bdd3344233f19232850">[6a64f65fe6]</a> Guido Falsi -- BuildSystem: Add patches to allow building with recent LibreSSL</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419">ASTERISK-26419</a>: audiohooks: Remove redundant codec translations when using audiohooks<br/>Reported by: Michael Walton<ul>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26230">ASTERISK-26230</a>: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f6a9617eb44a8d59b5828cd860d3852cc824ce9">[0f6a9617eb]</a> Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59c9bbe6961a5677ddb13eed2a130d16b6ffc0ee">[59c9bbe696]</a> Alexei Gradinari -- res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled</li>
-</ul><br><h
r><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27065">ASTERISK-27065</a>: call hangup after leaving app_queue<br/>Reported by: Marek Cervenka<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
-</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26469">ASTERISK-26469</a>: Infinite loop after a dual Redirect<br/>Reported by: Etienne Allovon<ul>
+</ul><br><h
4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26976">ASTERISK-26976</a>: libsrtp-2.x.x support<br/>Reported by: Alex<ul>
+<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</li>
+</ul><br><h
r><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26469">ASTERISK-26469</a>: Infinite loop after a dual Redirect<br/>Reported by: Etienne Allovon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b07b2162359ccc9a3f84324fabce18b6ad63eee3">[b07b216235]</a> Joshua Colp -- manager: Clear the flag on the other channel.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27095">ASTERISK-27095</a>: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bd7c0f37cb7b513d1333717ece0118bd8875546">[6bd7c0f37c]</a> George Joseph -- chan_pjsip: Fix ability to send UPDATE on COLP</li>
-</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27106">ASTERISK-27106</a>: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39d2ebbf56635355432eb96ff850c0c9bf2a5d63">[39d2ebbf56]</a> Alexander Traud -- chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9f4b3b966e911fae157a484d8f4a1440130eede6">[9f4b3b966e]</a> Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27016">ASTERISK-27016</a>: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.<br/>Reported by: Chris Howard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4910a3bf402baddf8ed72badfaed7ae64da48686">[4910a3bf40]</a> Joshua Colp -- channel: Fix reference counting in ast_channel_suppress.</li>
-</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27100">ASTERISK-27100</a>: channel: ast_waitfordigit_full fails to clear flag in an error branch.<br/>Reported by: Corey Farrell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73520e9f58857049a086fb88106e342cdc25d3a1">[73520e9f58]</a> Corey Farrell -- channel: Clear channel flag in error branch.</li>
-</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26978">ASTERISK-26978</a>: rtp: Crash in ast_rtp_codecs_payload_code()<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb48e99bd4f4556424a6799e2e5f7aebf8911e8d">[eb48e99bd4]</a> George Joseph -- bridge_native_rtp: Keep rtp instance refs on bridge_channel</li>
-</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27108">ASTERISK-27108</a>: Crash using 'data get' CLI command<br/>Reported by: Sean Bright<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6258de458b2e6ba02e91ed67bbd2801f0984526a">[6258de458b]</a> Sean Bright -- core: Fix segfault when invoking 'data get' CLI command</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27088">ASTERISK-27088</a>: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation<br/>Reported by: Joshua Colp<ul>
+</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27088">ASTERISK-27088</a>: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0426b1d88ab97c4fc1b2b27f8da93b28096f2dfc">[0426b1d88a]</a> Joshua Colp -- res_rtp_asterisk: Fix issues with ICE renegotiation.</li>
</ul><br><h4>Category: Resources/res_corosync</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25370">ASTERISK-25370</a>: res_corosync segfaults at startup with corosync version > 2.x<br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=005a4afa6b0e710e11b47b11cfc152b028c596fc">[005a4afa6b]</a> Jan Friesse -- res_corosync: Change thread stack size</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27090">ASTERISK-27090</a>: PJSIP: Deadlock using TCP transport<br/>Reported by: Richard Mudgett<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d64cbde5756eaa1c7ee62116e112b7ebd198bbe">[0d64cbde57]</a> Richard Mudgett -- pjsip_distributor.c: Fix deadlock with TCP type transports.</li>
</ul><br><h4>Category: Resources/res_pjsip_dialog_info_body_generator</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26919">ASTERISK-26919</a>: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip<br/>Reported by: Zach R<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6e4899612ca71bc3c9180dadea0c0117e8ae462">[a6e4899612]</a> Alexei Gradinari -- res_pjsip: New endpoint option "notify_early_inuse_ringing"</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27051">ASTERISK-27051</a>: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fbc34d2bd5393d93d8b3b3a8c6daa895c2e9633">[9fbc34d2bd]</a> Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode</li>
</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
+<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c00ee754b436ca926b92b469ce259e8fdc8732e">0c00ee754b</a></td><td>George Joseph</td><td>Update for 13.17.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=379fe658312e11699ff8c8e8a463e31b3c277237">379fe65831</a></td><td>George Joseph</td><td>Fix alembic branches</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=905d18e8bf52ea7657acaaf2ec0cbe58531fb625">905d18e8bf</a></td><td>Richard Mudgett</td><td>pjsip_distributor.c: Fix unidentified_requests hash functions.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f59d08924bc676970cabc6f3e291c7d1d2f2707">1f59d08924</a></td><td>Torrey Searle</td><td>res/res_pjsip_t38: fix incorrect increment of media_count</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c093bf8072ff65bf29d290c1330291c460cd7fdf">c093bf8072</a></td><td>Sean Bright</td><td>res_rtp_multicast: Use consistent timestamps when possible</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c10341646d353922b4ee92c77fc4e5560d263c73">c10341646d</a></td><td>George Joseph</td><td>test_json: Fix test names with reserved words</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=65898c3af82e2d780a48d9d50d3b1c952c208a89">65898c3af8</a></td><td>George Joseph</td><td>unittests: Add a unit test that causes a SEGV and...</td></tr>
-</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>.lastclean | 1
-.version | 1
-ChangeLog |51038 ----------
-asterisk-13.16.0-summary.html | 405
-asterisk-13.16.0-summary.txt | 952
-b/CHANGES | 54
-b/Makefile | 3
-b/addons/Makefile | 10
-b/apps/app_chanspy.c | 16
-b/apps/app_confbridge.c | 79
-b/apps/app_dial.c | 6
-b/apps/app_disa.c | 10
-b/apps/app_dumpchan.c | 4
-b/apps/app_externalivr.c | 6
-b/apps/app_meetme.c | 2
-b/apps/app_queue.c | 109
-b/apps/app_voicemail.c | 80
-b/autoconf/ast_ext_lib.m4 | 36
-b/bridges/bridge_native_rtp.c | 677
-b/bridges/bridge_simple.c | 32
-b/channels/chan_pjsip.c | 68
-b/channels/chan_sip.c | 8
-b/channels/pjsip/dialplan_functions.c | 37
-b/configs/samples/cdr.conf.sample | 2
-b/configs/samples/codecs.conf.sample | 6
-b/configs/samples/pjsip.conf.sample | 20
-b/configs/samples/sip.conf.sample | 3
-b/configs/samples/voicemail.conf.sample | 3
-b/configure | 434
-b/configure.ac | 100
-b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
-b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
-b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
-b/formats/format_g729.c | 2
-b/include/asterisk/ari.h | 10
-b/include/asterisk/autoconfig.h.in | 3
-b/include/asterisk/bridge_channel.h | 2
-b/include/asterisk/bridge_channel_internal.h | 11
-b/include/asterisk/bridge_technology.h | 3
-b/include/asterisk/channel.h | 25
-b/include/asterisk/codec.h | 3
-b/include/asterisk/core_local.h | 37
-b/include/asterisk/format.h | 11
-b/include/asterisk/res_pjsip.h | 74
-b/include/asterisk/res_pjsip_presence_xml.h | 3
-b/include/asterisk/res_pjsip_session.h | 11
-b/include/asterisk/rtp_engine.h | 9
-b/include/asterisk/smoother.h | 1
-b/include/asterisk/test.h | 8
-b/main/autoservice.c | 2
-b/main/bridge.c | 10
-b/main/bridge_after.c | 2
-b/main/bridge_channel.c | 38
-b/main/channel.c | 90
-b/main/codec_builtin.c | 19
-b/main/core_local.c | 54
-b/main/crypt.c | 2
-b/main/data.c | 4
-b/main/file.c | 20
-b/main/format.c | 8
-b/main/libasteriskssl.c | 4
-b/main/manager.c | 8
-b/main/pbx.c | 4
-b/main/pbx_app.c | 7
-b/main/pbx_builtins.c | 8
-b/main/tcptls.c | 4
-b/main/test.c | 4
-b/makeopts.in | 2
-b/res/res_agi.c | 73
-b/res/res_ari_applications.c | 4
-b/res/res_ari_asterisk.c | 4
-b/res/res_ari_bridges.c | 4
-b/res/res_ari_channels.c | 4
-b/res/res_ari_device_states.c | 4
-b/res/res_ari_endpoints.c | 4
-b/res/res_ari_events.c | 33
-b/res/res_ari_mailboxes.c | 4
-b/res/res_ari_playbacks.c | 4
-b/res/res_ari_recordings.c | 4
-b/res/res_ari_sounds.c | 4
-b/res/res_corosync.c | 29
-b/res/res_format_attr_h263.c | 2
-b/res/res_format_attr_h264.c | 2
-b/res/res_musiconhold.c | 4
-b/res/res_pjsip.c | 31
-b/res/res_pjsip/location.c | 53
-b/res/res_pjsip/pjsip_configuration.c | 9
-b/res/res_pjsip/pjsip_distributor.c | 242
-b/res/res_pjsip/presence_xml.c | 9
-b/res/res_pjsip_dialog_info_body_generator.c | 10
-b/res/res_pjsip_mwi.c | 87
-b/res/res_pjsip_pidf_body_generator.c | 2
-b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
-b/res/res_pjsip_pubsub.c | 8
-b/res/res_pjsip_refer.c | 28
-b/res/res_pjsip_sdp_rtp.c | 38
-b/res/res_pjsip_session.c | 37
-b/res/res_pjsip_session.exports.in | 1
-b/res/res_pjsip_t38.c | 2
-b/res/res_pjsip_transport_websocket.c | 4
-b/res/res_pjsip_xpidf_body_generator.c | 2
-b/res/res_rtp_asterisk.c | 41
-b/res/res_rtp_multicast.c | 139
-b/res/res_srtp.c | 15
-b/res/res_stasis.c | 20
-b/res/srtp/srtp_compat.h | 29
-b/res/stasis_recording/stored.c | 4
-b/rest-api-templates/res_ari_resource.c.mustache | 35
-b/tests/test_bridging.c | 292
-b/tests/test_json.c | 16
-b/tests/test_pbx.c | 22
-b/third-party/configure.m4 | 5
-b/third-party/pjproject/Makefile | 2
-b/third-party/pjproject/Makefile.rules | 7
-b/third-party/pjproject/configure.m4 | 6
-contrib/realtime/mssql/mssql_cdr.sql | 44
-contrib/realtime/mssql/mssql_config.sql | 1713
-contrib/realtime/mssql/mssql_voicemail.sql | 54
-contrib/realtime/mysql/mysql_cdr.sql | 32
-contrib/realtime/mysql/mysql_config.sql | 1052
-contrib/realtime/mysql/mysql_voicemail.sql | 34
-contrib/realtime/oracle/oracle_cdr.sql | 38
-contrib/realtime/oracle/oracle_config.sql | 1707
-contrib/realtime/oracle/oracle_voicemail.sql | 48
-contrib/realtime/postgresql/postgresql_cdr.sql | 36
-contrib/realtime/postgresql/postgresql_config.sql | 1130
-contrib/realtime/postgresql/postgresql_voicemail.sql | 38
-127 files changed, 3137 insertions(+), 58993 deletions(-)</pre><br></html>
\ No newline at end of file
+</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-13.16.0-summary.html | 405 ---
+asterisk-13.16.0-summary.txt | 952 ---------
+b/.version | 2
+b/CHANGES | 54
+b/ChangeLog | 1045 +++++++++-
+b/Makefile | 3
+b/addons/Makefile | 10
+b/apps/app_chanspy.c | 16
+b/apps/app_confbridge.c | 79
+b/apps/app_dial.c | 6
+b/apps/app_disa.c | 10
+b/apps/app_dumpchan.c | 4
+b/apps/app_externalivr.c | 6
+b/apps/app_meetme.c | 2
+b/apps/app_queue.c | 109 -
+b/apps/app_voicemail.c | 80
+b/asterisk-13.17.0-rc1-summary.html | 311 ++
+b/asterisk-13.17.0-rc1-summary.txt | 832 +++++++
+b/autoconf/ast_ext_lib.m4 | 36
+b/bridges/bridge_native_rtp.c | 677 +++++-
+b/bridges/bridge_simple.c | 32
+b/channels/chan_pjsip.c | 68
+b/channels/chan_sip.c | 8
+b/channels/pjsip/dialplan_functions.c | 37
+b/configs/samples/cdr.conf.sample | 2
+b/configs/samples/codecs.conf.sample | 6
+b/configs/samples/pjsip.conf.sample | 20
+b/configs/samples/sip.conf.sample | 3
+b/configs/samples/voicemail.conf.sample | 3
+b/configure | 434 +++-
+b/configure.ac | 100
+b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
+b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
+b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
+b/contrib/realtime/mssql/mssql_config.sql | 46
+b/contrib/realtime/mysql/mysql_config.sql | 18
+b/contrib/realtime/oracle/oracle_config.sql | 46
+b/contrib/realtime/postgresql/postgresql_config.sql | 22
+b/formats/format_g729.c | 2
+b/include/asterisk/ari.h | 10
+b/include/asterisk/autoconfig.h.in | 3
+b/include/asterisk/bridge_channel.h | 2
+b/include/asterisk/bridge_channel_internal.h | 11
+b/include/asterisk/bridge_technology.h | 3
+b/include/asterisk/channel.h | 25
+b/include/asterisk/codec.h | 3
+b/include/asterisk/core_local.h | 37
+b/include/asterisk/format.h | 11
+b/include/asterisk/res_pjsip.h | 74
+b/include/asterisk/res_pjsip_presence_xml.h | 3
+b/include/asterisk/res_pjsip_session.h | 11
+b/include/asterisk/rtp_engine.h | 9
+b/include/asterisk/smoother.h | 1
+b/include/asterisk/test.h | 8
+b/main/autoservice.c | 2
+b/main/bridge.c | 10
+b/main/bridge_after.c | 2
+b/main/bridge_channel.c | 38
+b/main/channel.c | 90
+b/main/codec_builtin.c | 19
+b/main/core_local.c | 54
+b/main/crypt.c | 2
+b/main/data.c | 4
+b/main/file.c | 20
+b/main/format.c | 8
+b/main/libasteriskssl.c | 4
+b/main/manager.c | 8
+b/main/pbx.c | 4
+b/main/pbx_app.c | 7
+b/main/pbx_builtins.c | 8
+b/main/tcptls.c | 4
+b/main/test.c | 4
+b/makeopts.in | 2
+b/res/res_agi.c | 73
+b/res/res_ari_applications.c | 4
+b/res/res_ari_asterisk.c | 4
+b/res/res_ari_bridges.c | 4
+b/res/res_ari_channels.c | 4
+b/res/res_ari_device_states.c | 4
+b/res/res_ari_endpoints.c | 4
+b/res/res_ari_events.c | 33
+b/res/res_ari_mailboxes.c | 4
+b/res/res_ari_playbacks.c | 4
+b/res/res_ari_recordings.c | 4
+b/res/res_ari_sounds.c | 4
+b/res/res_corosync.c | 29
+b/res/res_format_attr_h263.c | 2
+b/res/res_format_attr_h264.c | 2
+b/res/res_musiconhold.c | 4
+b/res/res_pjsip.c | 31
+b/res/res_pjsip/location.c | 53
+b/res/res_pjsip/pjsip_configuration.c | 9
+b/res/res_pjsip/pjsip_distributor.c | 242 +-
+b/res/res_pjsip/presence_xml.c | 9
+b/res/res_pjsip_dialog_info_body_generator.c | 10
+b/res/res_pjsip_mwi.c | 87
+b/res/res_pjsip_pidf_body_generator.c | 2
+b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
+b/res/res_pjsip_pubsub.c | 8
+b/res/res_pjsip_refer.c | 28
+b/res/res_pjsip_sdp_rtp.c | 38
+b/res/res_pjsip_session.c | 37
+b/res/res_pjsip_session.exports.in | 1
+b/res/res_pjsip_t38.c | 2
+b/res/res_pjsip_transport_websocket.c | 4
+b/res/res_pjsip_xpidf_body_generator.c | 2
+b/res/res_rtp_asterisk.c | 41
+b/res/res_rtp_multicast.c | 139 +
+b/res/res_srtp.c | 15
+b/res/res_stasis.c | 20
+b/res/srtp/srtp_compat.h | 29
+b/res/stasis_recording/stored.c | 4
+b/rest-api-templates/res_ari_resource.c.mustache | 35
+b/tests/test_bridging.c | 292 ++
+b/tests/test_json.c | 16
+b/tests/test_pbx.c | 22
+b/third-party/configure.m4 | 5
+b/third-party/pjproject/Makefile | 2
+b/third-party/pjproject/Makefile.rules | 7
+b/third-party/pjproject/configure.m4 | 24
+b/third-party/pjproject/patches/0070-Set-PJSIP_INV_SUPPORT_UPDATE-correctly-in-pjsip_inv_.patch | 16
+121 files changed, 5477 insertions(+), 2043 deletions(-)</pre><br></html>
\ No newline at end of file
Release Summary
- asterisk-13.17.0-rc1
+ asterisk-13.17.0
- Date: 2017-07-06
+ Date: 2017-07-12
<asteriskteam@digium.com>
Coders Testers Reporters
17 Sean Bright 4 Alexei Gradinari
- 11 George Joseph 4 Joshua Colp
+ 12 George Joseph 4 Joshua Colp
10 Joshua Colp 3 Kevin Harwell
9 Alexei Gradinari 3 Louis Jocelyn Paquet
5 Richard Mudgett 3 Tzafrir Cohen
Reported by: Ove Aursand
* [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
when a queue is empty
+ ASTERISK-27065: call hangup after leaving app_queue
+ Reported by: Marek Cervenka
+ * [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
+ when a queue is empty
ASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumasters
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
Category: Channels/chan_sip/General
+ ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really
+ different domain with TLS.
+ Reported by: Alexander Traud
+ * [39d2ebbf56] Alexander Traud -- chan_sip: Only when different, add
+ TCP|TLS in autodomain (SIP Domain Support).
+ * [9f4b3b966e] Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr
+ in autodomain (SIP Domain Support).
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngstrAP:m
Category: Core/Channels
+ ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an
+ error branch.
+ Reported by: Corey Farrell
+ * [73520e9f58] Corey Farrell -- channel: Clear channel flag in error
+ branch.
ASTERISK-27074: core_local: local channel data not being properly unref'ed
and unlocked
Reported by: Kevin Harwell
Category: Core/RTP
+ ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code()
+ Reported by: Ross Beer
+ * [eb48e99bd4] George Joseph -- bridge_native_rtp: Keep rtp instance
+ refs on bridge_channel
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
Category: General
+ ASTERISK-27108: Crash using 'data get' CLI command
+ Reported by: Sean Bright
+ * [6258de458b] Sean Bright -- core: Fix segfault when invoking 'data
+ get' CLI command
ASTERISK-27060: Comment typo format_g729.c
Reported by: Matthew Fredrickson
* [0a40073750] Matthew Fredrickson -- formats/format_g729: Fix typo in
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
+ Category: Resources/res_pjsip
+
+ ASTERISK-27090: PJSIP: Deadlock using TCP transport
+ Reported by: Richard Mudgett
+ * [0d64cbde57] Richard Mudgett -- pjsip_distributor.c: Fix deadlock with
+ TCP type transports.
+
Category: Resources/res_pjsip/Bundling
ASTERISK-27052: Asterisk build process fails with flag
* [6af2dd34af] Alexei Gradinari -- res_pjsip: New endpoint option
"refer_blind_progress"
- Information Request
-
- Category: Resources/res_rtp_asterisk
-
- ASTERISK-26976: libsrtp-2.x.x support
- Reported by: Alex
- * [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
-
Improvement
Category: Core/BuildSystem
* [59c9bbe696] Alexei Gradinari -- res_pjsip_mwi: don't create mwi
subscriptions if initial unsolicited disabled
+ Category: Resources/res_rtp_asterisk
+
+ ASTERISK-26976: libsrtp-2.x.x support
+ Reported by: Alex
+ * [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
+
----------------------------------------------------------------------
Open Issues
Bug
- Category: Applications/app_queue
-
- ASTERISK-27065: call hangup after leaving app_queue
- Reported by: Marek Cervenka
- * [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
- when a queue is empty
-
Category: Bridges/bridge_simple
ASTERISK-26469: Infinite loop after a dual Redirect
* [6bd7c0f37c] George Joseph -- chan_pjsip: Fix ability to send UPDATE
on COLP
- Category: Channels/chan_sip/General
-
- ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really
- different domain with TLS.
- Reported by: Alexander Traud
- * [39d2ebbf56] Alexander Traud -- chan_sip: Only when different, add
- TCP|TLS in autodomain (SIP Domain Support).
- * [9f4b3b966e] Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr
- in autodomain (SIP Domain Support).
-
Category: Core/Bridging
ASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events'
* [4910a3bf40] Joshua Colp -- channel: Fix reference counting in
ast_channel_suppress.
- Category: Core/Channels
-
- ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an
- error branch.
- Reported by: Corey Farrell
- * [73520e9f58] Corey Farrell -- channel: Clear channel flag in error
- branch.
-
- Category: Core/RTP
-
- ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code()
- Reported by: Ross Beer
- * [eb48e99bd4] George Joseph -- bridge_native_rtp: Keep rtp instance
- refs on bridge_channel
-
Category: General
- ASTERISK-27108: Crash using 'data get' CLI command
- Reported by: Sean Bright
- * [6258de458b] Sean Bright -- core: Fix segfault when invoking 'data
- get' CLI command
ASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and
unidirectional negotiation
Reported by: Joshua Colp
Reported by: mdu113
* [005a4afa6b] Jan Friesse -- res_corosync: Change thread stack size
- Category: Resources/res_pjsip
-
- ASTERISK-27090: PJSIP: Deadlock using TCP transport
- Reported by: Richard Mudgett
- * [0d64cbde57] Richard Mudgett -- pjsip_distributor.c: Fix deadlock with
- TCP type transports.
-
Category: Resources/res_pjsip_dialog_info_body_generator
ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+------------------+----------------------------------------|
+ | 0c00ee754b | George Joseph | Update for 13.17.0-rc1 |
+ |------------+------------------+----------------------------------------|
| 379fe65831 | George Joseph | Fix alembic branches |
|------------+------------------+----------------------------------------|
| 905d18e8bf | Richard Mudgett | pjsip_distributor.c: Fix |
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
- .lastclean | 1
- .version | 1
- ChangeLog |51038 ----------
- asterisk-13.16.0-summary.html | 405
- asterisk-13.16.0-summary.txt | 952
- b/CHANGES | 54
- b/Makefile | 3
- b/addons/Makefile | 10
- b/apps/app_chanspy.c | 16
- b/apps/app_confbridge.c | 79
- b/apps/app_dial.c | 6
- b/apps/app_disa.c | 10
- b/apps/app_dumpchan.c | 4
- b/apps/app_externalivr.c | 6
- b/apps/app_meetme.c | 2
- b/apps/app_queue.c | 109
- b/apps/app_voicemail.c | 80
- b/autoconf/ast_ext_lib.m4 | 36
- b/bridges/bridge_native_rtp.c | 677
- b/bridges/bridge_simple.c | 32
- b/channels/chan_pjsip.c | 68
- b/channels/chan_sip.c | 8
- b/channels/pjsip/dialplan_functions.c | 37
- b/configs/samples/cdr.conf.sample | 2
- b/configs/samples/codecs.conf.sample | 6
- b/configs/samples/pjsip.conf.sample | 20
- b/configs/samples/sip.conf.sample | 3
- b/configs/samples/voicemail.conf.sample | 3
- b/configure | 434
- b/configure.ac | 100
- b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
- b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
- b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
- b/formats/format_g729.c | 2
- b/include/asterisk/ari.h | 10
- b/include/asterisk/autoconfig.h.in | 3
- b/include/asterisk/bridge_channel.h | 2
- b/include/asterisk/bridge_channel_internal.h | 11
- b/include/asterisk/bridge_technology.h | 3
- b/include/asterisk/channel.h | 25
- b/include/asterisk/codec.h | 3
- b/include/asterisk/core_local.h | 37
- b/include/asterisk/format.h | 11
- b/include/asterisk/res_pjsip.h | 74
- b/include/asterisk/res_pjsip_presence_xml.h | 3
- b/include/asterisk/res_pjsip_session.h | 11
- b/include/asterisk/rtp_engine.h | 9
- b/include/asterisk/smoother.h | 1
- b/include/asterisk/test.h | 8
- b/main/autoservice.c | 2
- b/main/bridge.c | 10
- b/main/bridge_after.c | 2
- b/main/bridge_channel.c | 38
- b/main/channel.c | 90
- b/main/codec_builtin.c | 19
- b/main/core_local.c | 54
- b/main/crypt.c | 2
- b/main/data.c | 4
- b/main/file.c | 20
- b/main/format.c | 8
- b/main/libasteriskssl.c | 4
- b/main/manager.c | 8
- b/main/pbx.c | 4
- b/main/pbx_app.c | 7
- b/main/pbx_builtins.c | 8
- b/main/tcptls.c | 4
- b/main/test.c | 4
- b/makeopts.in | 2
- b/res/res_agi.c | 73
- b/res/res_ari_applications.c | 4
- b/res/res_ari_asterisk.c | 4
- b/res/res_ari_bridges.c | 4
- b/res/res_ari_channels.c | 4
- b/res/res_ari_device_states.c | 4
- b/res/res_ari_endpoints.c | 4
- b/res/res_ari_events.c | 33
- b/res/res_ari_mailboxes.c | 4
- b/res/res_ari_playbacks.c | 4
- b/res/res_ari_recordings.c | 4
- b/res/res_ari_sounds.c | 4
- b/res/res_corosync.c | 29
- b/res/res_format_attr_h263.c | 2
- b/res/res_format_attr_h264.c | 2
- b/res/res_musiconhold.c | 4
- b/res/res_pjsip.c | 31
- b/res/res_pjsip/location.c | 53
- b/res/res_pjsip/pjsip_configuration.c | 9
- b/res/res_pjsip/pjsip_distributor.c | 242
- b/res/res_pjsip/presence_xml.c | 9
- b/res/res_pjsip_dialog_info_body_generator.c | 10
- b/res/res_pjsip_mwi.c | 87
- b/res/res_pjsip_pidf_body_generator.c | 2
- b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
- b/res/res_pjsip_pubsub.c | 8
- b/res/res_pjsip_refer.c | 28
- b/res/res_pjsip_sdp_rtp.c | 38
- b/res/res_pjsip_session.c | 37
- b/res/res_pjsip_session.exports.in | 1
- b/res/res_pjsip_t38.c | 2
- b/res/res_pjsip_transport_websocket.c | 4
- b/res/res_pjsip_xpidf_body_generator.c | 2
- b/res/res_rtp_asterisk.c | 41
- b/res/res_rtp_multicast.c | 139
- b/res/res_srtp.c | 15
- b/res/res_stasis.c | 20
- b/res/srtp/srtp_compat.h | 29
- b/res/stasis_recording/stored.c | 4
- b/rest-api-templates/res_ari_resource.c.mustache | 35
- b/tests/test_bridging.c | 292
- b/tests/test_json.c | 16
- b/tests/test_pbx.c | 22
- b/third-party/configure.m4 | 5
- b/third-party/pjproject/Makefile | 2
- b/third-party/pjproject/Makefile.rules | 7
- b/third-party/pjproject/configure.m4 | 6
- contrib/realtime/mssql/mssql_cdr.sql | 44
- contrib/realtime/mssql/mssql_config.sql | 1713
- contrib/realtime/mssql/mssql_voicemail.sql | 54
- contrib/realtime/mysql/mysql_cdr.sql | 32
- contrib/realtime/mysql/mysql_config.sql | 1052
- contrib/realtime/mysql/mysql_voicemail.sql | 34
- contrib/realtime/oracle/oracle_cdr.sql | 38
- contrib/realtime/oracle/oracle_config.sql | 1707
- contrib/realtime/oracle/oracle_voicemail.sql | 48
- contrib/realtime/postgresql/postgresql_cdr.sql | 36
- contrib/realtime/postgresql/postgresql_config.sql | 1130
- contrib/realtime/postgresql/postgresql_voicemail.sql | 38
- 127 files changed, 3137 insertions(+), 58993 deletions(-)
+ asterisk-13.16.0-summary.html | 405 ---
+ asterisk-13.16.0-summary.txt | 952 ---------
+ b/.version | 2
+ b/CHANGES | 54
+ b/ChangeLog | 1045 +++++++++-
+ b/Makefile | 3
+ b/addons/Makefile | 10
+ b/apps/app_chanspy.c | 16
+ b/apps/app_confbridge.c | 79
+ b/apps/app_dial.c | 6
+ b/apps/app_disa.c | 10
+ b/apps/app_dumpchan.c | 4
+ b/apps/app_externalivr.c | 6
+ b/apps/app_meetme.c | 2
+ b/apps/app_queue.c | 109 -
+ b/apps/app_voicemail.c | 80
+ b/asterisk-13.17.0-rc1-summary.html | 311 ++
+ b/asterisk-13.17.0-rc1-summary.txt | 832 +++++++
+ b/autoconf/ast_ext_lib.m4 | 36
+ b/bridges/bridge_native_rtp.c | 677 +++++-
+ b/bridges/bridge_simple.c | 32
+ b/channels/chan_pjsip.c | 68
+ b/channels/chan_sip.c | 8
+ b/channels/pjsip/dialplan_functions.c | 37
+ b/configs/samples/cdr.conf.sample | 2
+ b/configs/samples/codecs.conf.sample | 6
+ b/configs/samples/pjsip.conf.sample | 20
+ b/configs/samples/sip.conf.sample | 3
+ b/configs/samples/voicemail.conf.sample | 3
+ b/configure | 434 +++-
+ b/configure.ac | 100
+ b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
+ b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
+ b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
+ b/contrib/realtime/mssql/mssql_config.sql | 46
+ b/contrib/realtime/mysql/mysql_config.sql | 18
+ b/contrib/realtime/oracle/oracle_config.sql | 46
+ b/contrib/realtime/postgresql/postgresql_config.sql | 22
+ b/formats/format_g729.c | 2
+ b/include/asterisk/ari.h | 10
+ b/include/asterisk/autoconfig.h.in | 3
+ b/include/asterisk/bridge_channel.h | 2
+ b/include/asterisk/bridge_channel_internal.h | 11
+ b/include/asterisk/bridge_technology.h | 3
+ b/include/asterisk/channel.h | 25
+ b/include/asterisk/codec.h | 3
+ b/include/asterisk/core_local.h | 37
+ b/include/asterisk/format.h | 11
+ b/include/asterisk/res_pjsip.h | 74
+ b/include/asterisk/res_pjsip_presence_xml.h | 3
+ b/include/asterisk/res_pjsip_session.h | 11
+ b/include/asterisk/rtp_engine.h | 9
+ b/include/asterisk/smoother.h | 1
+ b/include/asterisk/test.h | 8
+ b/main/autoservice.c | 2
+ b/main/bridge.c | 10
+ b/main/bridge_after.c | 2
+ b/main/bridge_channel.c | 38
+ b/main/channel.c | 90
+ b/main/codec_builtin.c | 19
+ b/main/core_local.c | 54
+ b/main/crypt.c | 2
+ b/main/data.c | 4
+ b/main/file.c | 20
+ b/main/format.c | 8
+ b/main/libasteriskssl.c | 4
+ b/main/manager.c | 8
+ b/main/pbx.c | 4
+ b/main/pbx_app.c | 7
+ b/main/pbx_builtins.c | 8
+ b/main/tcptls.c | 4
+ b/main/test.c | 4
+ b/makeopts.in | 2
+ b/res/res_agi.c | 73
+ b/res/res_ari_applications.c | 4
+ b/res/res_ari_asterisk.c | 4
+ b/res/res_ari_bridges.c | 4
+ b/res/res_ari_channels.c | 4
+ b/res/res_ari_device_states.c | 4
+ b/res/res_ari_endpoints.c | 4
+ b/res/res_ari_events.c | 33
+ b/res/res_ari_mailboxes.c | 4
+ b/res/res_ari_playbacks.c | 4
+ b/res/res_ari_recordings.c | 4
+ b/res/res_ari_sounds.c | 4
+ b/res/res_corosync.c | 29
+ b/res/res_format_attr_h263.c | 2
+ b/res/res_format_attr_h264.c | 2
+ b/res/res_musiconhold.c | 4
+ b/res/res_pjsip.c | 31
+ b/res/res_pjsip/location.c | 53
+ b/res/res_pjsip/pjsip_configuration.c | 9
+ b/res/res_pjsip/pjsip_distributor.c | 242 +-
+ b/res/res_pjsip/presence_xml.c | 9
+ b/res/res_pjsip_dialog_info_body_generator.c | 10
+ b/res/res_pjsip_mwi.c | 87
+ b/res/res_pjsip_pidf_body_generator.c | 2
+ b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
+ b/res/res_pjsip_pubsub.c | 8
+ b/res/res_pjsip_refer.c | 28
+ b/res/res_pjsip_sdp_rtp.c | 38
+ b/res/res_pjsip_session.c | 37
+ b/res/res_pjsip_session.exports.in | 1
+ b/res/res_pjsip_t38.c | 2
+ b/res/res_pjsip_transport_websocket.c | 4
+ b/res/res_pjsip_xpidf_body_generator.c | 2
+ b/res/res_rtp_asterisk.c | 41
+ b/res/res_rtp_multicast.c | 139 +
+ b/res/res_srtp.c | 15
+ b/res/res_stasis.c | 20
+ b/res/srtp/srtp_compat.h | 29
+ b/res/stasis_recording/stored.c | 4
+ b/rest-api-templates/res_ari_resource.c.mustache | 35
+ b/tests/test_bridging.c | 292 ++
+ b/tests/test_json.c | 16
+ b/tests/test_pbx.c | 22
+ b/third-party/configure.m4 | 5
+ b/third-party/pjproject/Makefile | 2
+ b/third-party/pjproject/Makefile.rules | 7
+ b/third-party/pjproject/configure.m4 | 24
+ b/third-party/pjproject/patches/0070-Set-PJSIP_INV_SUPPORT_UPDATE-correctly-in-pjsip_inv_.patch | 16
+ 121 files changed, 5477 insertions(+), 2043 deletions(-)
projects / thirdparty / asterisk.git / commitdiff
