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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><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>
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<li><a href="#summary">Summary</a></li>
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<li><a href="#contributors">Contributors</a></li>
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<li><a href="#closed_issues">Closed Issues</a></li>
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<li><a href="#open_issues">Open Issues</a></li>
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<li><a href="#commits">Other Changes</a></li>
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<li><a href="#diffstat">Diffstat</a></li>
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</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">
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<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
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<tr valign="top"><td width="33%">17 Sean Bright <sean.bright@gmail.com><br/>11 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>
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</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>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97b003f5e2d4a350508fc20173e180a23f8ef525">[97b003f5e2]</a> Sean Bright -- format_mp3: Re-work menuselect/build issues</li>
|
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<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>
|
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</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27012">ASTERISK-27012</a>: app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br/>Reported by: Robert Mordec<ul>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1b32de2c5fb8854183f0c7d8c9df7470ab9c140">[f1b32de2c5]</a> Robert Mordec -- app_confbridge: Race between removing and playing name recording while leaving</li>
|
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</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
|
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</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-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=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
|
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</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26715">ASTERISK-26715</a>: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel<br/>Reported by: David Brillert<ul>
|
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<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>
|
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</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26975">ASTERISK-26975</a>: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call<br/>Reported by: Lorne Gaetz<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>
|
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</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24052">ASTERISK-24052</a>: app_voicemail reloads result in leaked IMAP sockets.<br/>Reported by: Louis Jocelyn Paquet<ul>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b6c327c515944d74aa798f385e01768a4bb04c2">[3b6c327c51]</a> Alexei Gradinari -- app_voicemail: IMAP logout on reload/unload</li>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08be5e01e8ab72a7e9e80525e20967467a6df99b">[08be5e01e8]</a> Alexei Gradinari -- app_voicemail: IMAP logout on MWI unsubscribe</li>
|
||||
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<ul>
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<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>
|
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</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27039">ASTERISK-27039</a>: chan_pjsip: Device state is idle when channel from endpoint is in early media<br/>Reported by: Joshua Colp<ul>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f10c6b3b044f9979c523f65f449670047dcb57f">[1f10c6b3b0]</a> Joshua Colp -- chan_pjsip: Update device state when in early media.</li>
|
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</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26996">ASTERISK-26996</a>: chan_pjsip: Flipping between codecs<br/>Reported by: Michael Maier<ul>
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<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>
|
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</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>
|
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<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>
|
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</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>
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<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>
|
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</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>
|
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<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>
|
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</ul><br><h4>Category: Codecs/General</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>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
|
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</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27075">ASTERISK-27075</a>: bridge: stuck channel(s) after failed attended transfer<br/>Reported by: Kevin Harwell<ul>
|
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<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>
|
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</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>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
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</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>
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<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>
|
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</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>
|
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
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</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
|
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</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26789">ASTERISK-26789</a>: Audit manipulation of channel flags without locks<br/>Reported by: Joshua Colp<ul>
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<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>
|
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</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>
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<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>
|
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</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>
|
||||
<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>
|
||||
<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>
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||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=283cc59af746896a2b2bc23899fc86118895f7c0">[283cc59af7]</a> Sean Bright -- pbx_builtin: Properly handle hangup during Background</li>
|
||||
</ul><br><h4>Category: Resources/res_agi</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><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22432">ASTERISK-22432</a>: Async AGI crashes Asterisk when issuing "set variable" command without args<br/>Reported by: Antoine Pitrou<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f306e451f6f905a2bb74c15cb844735c244a7610">[f306e451f6]</a> Sean Bright -- res_agi: Prevent crash when SET VARIABLE called without arguments</li>
|
||||
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25662">ASTERISK-25662</a>: Malformed AGI 520 Usage response<br/>Reported by: Tony Mountifield<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a007e438c36960d4179e2f188767e7ae14a204d1">[a007e438c3]</a> Sean Bright -- res_agi: Fix malformed AGI usage response</li>
|
||||
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27026">ASTERISK-27026</a>: res_ari: Crash when no ari.conf configuration file exists<br/>Reported by: Ronald Raikes<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7901b9853e8f60e1d2dce44ce81dec6f7f866ccc">[7901b9853e]</a> George Joseph -- res_ari: Add "module loaded" check to ari stubs</li>
|
||||
</ul><br><h4>Category: Resources/res_ari_recordings</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27021">ASTERISK-27021</a>: GET /recordings/stored returns 500 Internal Server Error<br/>Reported by: Tim Morgan<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf6cf59646f52dc3de12dac16c3c3824ce9ae927">[cf6cf59646]</a> Sean Bright -- stasis_recording: Correct ast_asprintf error checking</li>
|
||||
</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27008">ASTERISK-27008</a>: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br/>Reported by: John Harris<ul>
|
||||
<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/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=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
|
||||
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27024">ASTERISK-27024</a>: nat/external_media settings ignored in 14.4.1<br/>Reported by: Christopher van de Sande<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2dee95cc7a280d0ab84c778bf44a76aa62ac758b">[2dee95cc7a]</a> Florian Floimair -- res_pjsip_session: Correct inverted test in session_outgoing_nat_hook</li>
|
||||
</ul><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=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
|
||||
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26964">ASTERISK-26964</a>: res_pjsip_session: Wrong From on reinvite when request and To URI differ<br/>Reported by: Yasin CANER<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36628cc9c474b52b134a415803b14f87e420dce6">[36628cc9c4]</a> Yasin CANER -- res_pjsip_session : fixed wrong From Header number On Re-invite</li>
|
||||
</ul><br><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27046">ASTERISK-27046</a>: res_pjsip_transport_websocket: segfault in get_write_timeout<br/>Reported by: Jørgen H<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e16a669c70c5a93bb9a38c218a5348cd62bd780a">[e16a669c70]</a> Jørgen H -- res_pjsip_transport_websocket: Add NULL check in get_write_timeout</li>
|
||||
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27022">ASTERISK-27022</a>: res_rtp_asterisk: Incorrect SSRC change for RTCP component<br/>Reported by: Michael Walton<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7dafe82751fd512d58bb3843601daff013958dd2">[7dafe82751]</a> George Joseph -- res_rtp_asterisk: Fix ssrc change for rtcp srtp</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><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>
|
||||
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</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: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25294">ASTERISK-25294</a>: srtp's crypto_get_random deprecated<br/>Reported by: Tzafrir Cohen<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><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>
|
||||
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
|
||||
</ul><br><h4>Category: Resources/res_stasis_snoop</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<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>
|
||||
<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: Core/General</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>
|
||||
<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: Core/Portability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27042">ASTERISK-27042</a>: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br/>Reported by: Guido Falsi<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44cee2f4a15db911d2c9bdd6f845d17a1e6c6c17">[44cee2f4a1]</a> Guido Falsi -- BuildSystem: Fix build on FreeBSD due to missing crypt.h</li>
|
||||
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26124">ASTERISK-26124</a>: res_agi: Set audio format for EAGI audio stream<br/>Reported by: John Fawcett<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90237dca11d0adf129198cef4a6a0716a52618b5">[90237dca11]</a> Sean Bright -- res_agi: Allow configuration of audio format of EAGI pipe</li>
|
||||
</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><hr><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>
|
||||
<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>
|
||||
<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=8e749c8f51c20fb13bfe93e969cf02d7e74cdb27">[8e749c8f51]</a> Alexei Gradinari -- res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact</li>
|
||||
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27059">ASTERISK-27059</a>: res_stasis: Stolen channel references are leaking<br/>Reported by: George Joseph<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=edfdb4dff5d8438bdb1dfb526c57618944ea6bf3">[edfdb4dff5]</a> George Joseph -- res_stasis: Plug reference leak on stolen channels</li>
|
||||
</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27097">ASTERISK-27097</a>: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure<br/>Reported by: George Joseph<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbe68f139db525b2d922f63d8452d9732fb5f1b9">[bbe68f139d]</a> George Joseph -- pjproject_bundled: Allow passing configure options to bundled</li>
|
||||
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27068">ASTERISK-27068</a>: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect<br/>Reported by: Alexei Gradinari<ul>
|
||||
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
|
||||
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<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><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<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=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=764d04fa8705d9e5c2e7aee8a6f1c774d7d28595">764d04fa87</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cecf6540dc4779598289f711340bb966bbfcc6aa">cecf6540dc</a></td><td>Rodrigo Ramírez Norambuena</td><td>cdr: fix mistake spelling of a word for Unanswered.</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9a4ab8c8c00c8d53584d6f7e31729b5027c8dd6">b9a4ab8c8c</a></td><td>Richard Mudgett</td><td>chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read.</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1a209d5ac8f8b7fe96e54d6aba55dbf0dbb1403">f1a209d5ac</a></td><td>Richard Mudgett</td><td>app_voicemail.c: Fix compile error when IMAP enabled.</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68de35a6a01e2a1fe732e156b73f800bb672a421">68de35a6a0</a></td><td>David M. Lee</td><td>CFLAGS for BIND8 support</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da3312457e6cf1c0d7bc8cb2a4aba57877fb5afc">da3312457e</a></td><td>Sean Bright</td><td>codecs.conf.sample: Fix max_bandwidth speling error</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=590ffcaf0b03bbe3d25730ad750a2075a46c7208">590ffcaf0b</a></td><td>Sean Bright</td><td>eventfd: Disable during cross compilation</td></tr>
|
||||
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5520b6c201875133a73db5a2c88b5fc5b78864bb">5520b6c201</a></td><td>Alexei Gradinari</td><td>CHANGES: correct version for a new option 'refer_blind_progress'</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>
|
||||
832
asterisk-13.17.0-rc1-summary.txt
Normal file
832
asterisk-13.17.0-rc1-summary.txt
Normal file
@@ -0,0 +1,832 @@
|
||||
Release Summary
|
||||
|
||||
asterisk-13.17.0-rc1
|
||||
|
||||
Date: 2017-07-06
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Closed Issues
|
||||
4. Open Issues
|
||||
5. Other Changes
|
||||
6. Diffstat
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Summary
|
||||
|
||||
[Back to Top]
|
||||
|
||||
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.
|
||||
|
||||
The data in this summary reflects changes that have been made since the
|
||||
previous release, asterisk-13.16.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Contributors
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This table lists the people who have submitted code, those that have
|
||||
tested patches, as well as those that reported issues on the issue tracker
|
||||
that were resolved in this release. For coders, the number is how many of
|
||||
their patches (of any size) were committed into this release. For testers,
|
||||
the number is the number of times their name was listed as assisting with
|
||||
testing a patch. Finally, for reporters, the number is the number of
|
||||
issues that they reported that were affected by commits that went into
|
||||
this release.
|
||||
|
||||
Coders Testers Reporters
|
||||
17 Sean Bright 4 Alexei Gradinari
|
||||
11 George Joseph 4 Joshua Colp
|
||||
10 Joshua Colp 3 Kevin Harwell
|
||||
9 Alexei Gradinari 3 Louis Jocelyn Paquet
|
||||
5 Richard Mudgett 3 Tzafrir Cohen
|
||||
5 Kevin Harwell 3 George Joseph
|
||||
2 Torrey Searle 2 Guido Falsi
|
||||
2 Guido Falsi 2 Alexander Traud
|
||||
2 Alexander Traud 2 Michael Walton
|
||||
1 Jan Friesse 2 Torrey Searle
|
||||
1 Florian Floimair 1 Rusty Newton
|
||||
1 Ivan Poddubny 1 Matthew Fredrickson
|
||||
1 Matthew Fredrickson 1 Jacek Konieczny
|
||||
1 Yasin CANER 1 Tim Morgan
|
||||
1 David M. Lee 1 Etienne Allovon
|
||||
1 Robert Mordec 1 alex
|
||||
1 JA,rgen H 1 Kinsey Moore
|
||||
1 Rodrigo Ramirez Norambuena 1 John Harris
|
||||
1 Frederic LE FOLL 1 Javier Riveros
|
||||
1 Corey Farrell 1 Sean Bright
|
||||
1 Robert Mordec
|
||||
1 Ross Beer
|
||||
1 Chris Howard
|
||||
1 mdu113
|
||||
1 Andrew Nowrot
|
||||
1 'alex'
|
||||
1 Lorne Gaetz
|
||||
1 Ben Langfeld
|
||||
1 John Fawcett
|
||||
1 Corey Farrell
|
||||
1 Frankie Chin
|
||||
1 Zach R
|
||||
1 Matthias Binder
|
||||
1 Christopher van de Sande
|
||||
1 Stefan EngstrAP:m
|
||||
1 Antoine Pitrou
|
||||
1 Alex
|
||||
1 Etienne Lessard
|
||||
1 Ryan Smith
|
||||
1 Michael Maier
|
||||
1 OpenBSD ports
|
||||
1 Marek Cervenka
|
||||
1 Ronald Raikes
|
||||
1 Ove Aursand
|
||||
1 Richard Mudgett
|
||||
1 Frederic LE FOLL
|
||||
1 wushumasters
|
||||
1 Tony Mountifield
|
||||
1 JA,rgen H
|
||||
1 Michel R. Vaillancourt
|
||||
1 David Brillert
|
||||
1 Yasin CANER
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
Bug
|
||||
|
||||
Category: Addons/format_mp3
|
||||
|
||||
ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if
|
||||
mp3lib was not downloaded
|
||||
Reported by: Tzafrir Cohen
|
||||
* [97b003f5e2] Sean Bright -- format_mp3: Re-work menuselect/build
|
||||
issues
|
||||
* [72213c98e3] Sean Bright -- format_mp3: Don't try to build format_mp3
|
||||
if we don't have sources
|
||||
|
||||
Category: Applications/app_confbridge
|
||||
|
||||
ASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user
|
||||
name recording while leaving
|
||||
Reported by: Robert Mordec
|
||||
* [f1b32de2c5] Robert Mordec -- app_confbridge: Race between removing
|
||||
and playing name recording while leaving
|
||||
|
||||
Category: Applications/app_meetme
|
||||
|
||||
ASTERISK-27025: channel / meetme: Fix missing parentheses
|
||||
Reported by: Joshua Colp
|
||||
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
|
||||
|
||||
Category: Applications/app_queue
|
||||
|
||||
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
|
||||
Reported by: Ove Aursand
|
||||
* [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
|
||||
call when not.
|
||||
ASTERISK-26400: app_queue: Queue member stops being called after AMI
|
||||
"Redirect" action for queues with wrapuptime
|
||||
Reported by: Etienne Lessard
|
||||
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
|
||||
call when not.
|
||||
ASTERISK-26715: app_queue: Member will not receive any new calls after
|
||||
doing a transfer if wrapuptime = greater than 0 and using Local channel
|
||||
Reported by: David Brillert
|
||||
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
|
||||
call when not.
|
||||
ASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to
|
||||
receive queue calls after transfer queue call
|
||||
Reported by: Lorne Gaetz
|
||||
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
|
||||
call when not.
|
||||
|
||||
Category: Applications/app_voicemail/IMAP
|
||||
|
||||
ASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets.
|
||||
Reported by: Louis Jocelyn Paquet
|
||||
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
|
||||
control
|
||||
* [3b6c327c51] Alexei Gradinari -- app_voicemail: IMAP logout on
|
||||
reload/unload
|
||||
* [08be5e01e8] Alexei Gradinari -- app_voicemail: IMAP logout on MWI
|
||||
unsubscribe
|
||||
|
||||
Category: Bridges/bridge_simple
|
||||
|
||||
ASTERISK-26973: bridge: Crash when freeing frame and snooping
|
||||
Reported by: Michel R. Vaillancourt
|
||||
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
|
||||
after call to audiohooks
|
||||
|
||||
Category: Channels/chan_pjsip
|
||||
|
||||
ASTERISK-27039: chan_pjsip: Device state is idle when channel from
|
||||
endpoint is in early media
|
||||
Reported by: Joshua Colp
|
||||
* [1f10c6b3b0] Joshua Colp -- chan_pjsip: Update device state when in
|
||||
early media.
|
||||
ASTERISK-26996: chan_pjsip: Flipping between codecs
|
||||
Reported by: Michael Maier
|
||||
* [996a4791ff] Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec'
|
||||
option to include us changing.
|
||||
ASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints
|
||||
Reported by: Jacek Konieczny
|
||||
* [746c2c5745] Joshua Colp -- res_pjsip: Add support for returning only
|
||||
reachable contacts and use it.
|
||||
|
||||
Category: Channels/chan_sip/General
|
||||
|
||||
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
|
||||
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
|
||||
|
||||
Category: Channels/chan_sip/SRTP
|
||||
|
||||
ASTERISK-25101: DTLS configuration can not be specified in the general
|
||||
section - documentation
|
||||
Reported by: Ben Langfeld
|
||||
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
|
||||
settings are permitted
|
||||
|
||||
Category: Codecs/General
|
||||
|
||||
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
|
||||
order on Intel platform when using slin codec
|
||||
Reported by: Frankie Chin
|
||||
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
|
||||
|
||||
Category: Core/Bridging
|
||||
|
||||
ASTERISK-27075: bridge: stuck channel(s) after failed attended transfer
|
||||
Reported by: Kevin Harwell
|
||||
* [67664fbf95] Kevin Harwell -- bridge: stuck channel(s) after failed
|
||||
attended transfer
|
||||
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
|
||||
bridge
|
||||
Reported by: Torrey Searle
|
||||
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
|
||||
|
||||
Category: Core/Channels
|
||||
|
||||
ASTERISK-27074: core_local: local channel data not being properly unref'ed
|
||||
and unlocked
|
||||
Reported by: Kevin Harwell
|
||||
* [1f9913f272] Kevin Harwell -- core_local: local channel data not being
|
||||
properly unref'ed and unlocked
|
||||
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
|
||||
bridge
|
||||
Reported by: Torrey Searle
|
||||
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
|
||||
ASTERISK-27025: channel / meetme: Fix missing parentheses
|
||||
Reported by: Joshua Colp
|
||||
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
|
||||
|
||||
Category: Core/General
|
||||
|
||||
ASTERISK-26789: Audit manipulation of channel flags without locks
|
||||
Reported by: Joshua Colp
|
||||
* [1618203964] Joshua Colp -- asterisk: Audit locking of channel when
|
||||
manipulating flags.
|
||||
|
||||
Category: Core/PBX
|
||||
|
||||
ASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and
|
||||
application unregistration
|
||||
Reported by: Frederic LE FOLL
|
||||
* [dc307af7f2] Frederic LE FOLL -- Core/PBX: Deadlock between dialplan
|
||||
execution and application unregistration.
|
||||
|
||||
Category: Core/RTP
|
||||
|
||||
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
|
||||
order on Intel platform when using slin codec
|
||||
Reported by: Frankie Chin
|
||||
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
|
||||
|
||||
Category: Core/Sorcery
|
||||
|
||||
ASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c
|
||||
Reported by: Ryan Smith
|
||||
* [c2eea791e4] George Joseph -- res_pjsip_pubsub: Fix reference to
|
||||
released endpoint
|
||||
|
||||
Category: Documentation
|
||||
|
||||
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
|
||||
argument
|
||||
Reported by: Rusty Newton
|
||||
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
|
||||
documentation
|
||||
|
||||
Category: General
|
||||
|
||||
ASTERISK-27060: Comment typo format_g729.c
|
||||
Reported by: Matthew Fredrickson
|
||||
* [0a40073750] Matthew Fredrickson -- formats/format_g729: Fix typo in
|
||||
comment
|
||||
|
||||
Category: PBX/pbx_realtime
|
||||
|
||||
ASTERISK-19291: Background in realtime
|
||||
Reported by: Andrew Nowrot
|
||||
* [283cc59af7] Sean Bright -- pbx_builtin: Properly handle hangup during
|
||||
Background
|
||||
|
||||
Category: Resources/res_agi
|
||||
|
||||
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
|
||||
argument
|
||||
Reported by: Rusty Newton
|
||||
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
|
||||
documentation
|
||||
ASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable"
|
||||
command without args
|
||||
Reported by: Antoine Pitrou
|
||||
* [f306e451f6] Sean Bright -- res_agi: Prevent crash when SET VARIABLE
|
||||
called without arguments
|
||||
ASTERISK-25662: Malformed AGI 520 Usage response
|
||||
Reported by: Tony Mountifield
|
||||
* [a007e438c3] Sean Bright -- res_agi: Fix malformed AGI usage response
|
||||
|
||||
Category: Resources/res_ari
|
||||
|
||||
ASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists
|
||||
Reported by: Ronald Raikes
|
||||
* [7901b9853e] George Joseph -- res_ari: Add "module loaded" check to
|
||||
ari stubs
|
||||
|
||||
Category: Resources/res_ari_recordings
|
||||
|
||||
ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error
|
||||
Reported by: Tim Morgan
|
||||
* [cf6cf59646] Sean Bright -- stasis_recording: Correct ast_asprintf
|
||||
error checking
|
||||
|
||||
Category: Resources/res_format_attr_h264
|
||||
|
||||
ASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional
|
||||
parameters have a space
|
||||
Reported by: John Harris
|
||||
* [700ef6861a] Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp
|
||||
attributes
|
||||
|
||||
Category: Resources/res_parking
|
||||
|
||||
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
|
||||
call when not.
|
||||
|
||||
Category: Resources/res_pjsip/Bundling
|
||||
|
||||
ASTERISK-27052: Asterisk build process fails with flag
|
||||
--with-pjproject-bundled with curl download command and slow network
|
||||
Reported by: alex
|
||||
* [0bde568669] George Joseph -- pjproject_bundled: Use the asterisk
|
||||
github mirror for download
|
||||
|
||||
Category: Resources/res_pjsip_refer
|
||||
|
||||
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
|
||||
Reported by: Kevin Harwell
|
||||
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
|
||||
during transfer
|
||||
|
||||
Category: Resources/res_pjsip_session
|
||||
|
||||
ASTERISK-27024: nat/external_media settings ignored in 14.4.1
|
||||
Reported by: Christopher van de Sande
|
||||
* [2dee95cc7a] Florian Floimair -- res_pjsip_session: Correct inverted
|
||||
test in session_outgoing_nat_hook
|
||||
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
|
||||
Reported by: Kevin Harwell
|
||||
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
|
||||
during transfer
|
||||
ASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and
|
||||
To URI differ
|
||||
Reported by: Yasin CANER
|
||||
* [36628cc9c4] Yasin CANER -- res_pjsip_session : fixed wrong From
|
||||
Header number On Re-invite
|
||||
|
||||
Category: Resources/res_pjsip_transport_websocket
|
||||
|
||||
ASTERISK-27046: res_pjsip_transport_websocket: segfault in
|
||||
get_write_timeout
|
||||
Reported by: JA,rgen H
|
||||
* [e16a669c70] JA,rgen H -- res_pjsip_transport_websocket: Add NULL
|
||||
check in get_write_timeout
|
||||
|
||||
Category: Resources/res_rtp_asterisk
|
||||
|
||||
ASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component
|
||||
Reported by: Michael Walton
|
||||
* [7dafe82751] George Joseph -- res_rtp_asterisk: Fix ssrc change for
|
||||
rtcp srtp
|
||||
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
|
||||
order on Intel platform when using slin codec
|
||||
Reported by: Frankie Chin
|
||||
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
|
||||
ASTERISK-25101: DTLS configuration can not be specified in the general
|
||||
section - documentation
|
||||
Reported by: Ben Langfeld
|
||||
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
|
||||
settings are permitted
|
||||
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
|
||||
authentication failure 10 or 110
|
||||
Reported by: Javier Riveros
|
||||
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
|
||||
wrong srtp unprotecting algorithm
|
||||
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
|
||||
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
|
||||
|
||||
Category: Resources/res_srtp
|
||||
|
||||
ASTERISK-25294: srtp's crypto_get_random deprecated
|
||||
Reported by: Tzafrir Cohen
|
||||
* [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
|
||||
ASTERISK-25101: DTLS configuration can not be specified in the general
|
||||
section - documentation
|
||||
Reported by: Ben Langfeld
|
||||
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
|
||||
settings are permitted
|
||||
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
|
||||
authentication failure 10 or 110
|
||||
Reported by: Javier Riveros
|
||||
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
|
||||
wrong srtp unprotecting algorithm
|
||||
|
||||
Category: Resources/res_stasis_snoop
|
||||
|
||||
ASTERISK-26973: bridge: Crash when freeing frame and snooping
|
||||
Reported by: Michel R. Vaillancourt
|
||||
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
|
||||
after call to audiohooks
|
||||
|
||||
Category: pjproject/pjsip
|
||||
|
||||
ASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i)
|
||||
Reported by: Matthias Binder
|
||||
* [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
|
||||
|
||||
ASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL
|
||||
Reported by: Guido Falsi
|
||||
* [6a64f65fe6] Guido Falsi -- BuildSystem: Add patches to allow building
|
||||
with recent LibreSSL
|
||||
|
||||
Category: Core/Channels
|
||||
|
||||
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
|
||||
audiohooks
|
||||
Reported by: Michael Walton
|
||||
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
|
||||
after call to audiohooks
|
||||
|
||||
Category: Core/General
|
||||
|
||||
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
|
||||
audiohooks
|
||||
Reported by: Michael Walton
|
||||
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
|
||||
after call to audiohooks
|
||||
|
||||
Category: Core/Portability
|
||||
|
||||
ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to
|
||||
missing crypt.h file
|
||||
Reported by: Guido Falsi
|
||||
* [44cee2f4a1] Guido Falsi -- BuildSystem: Fix build on FreeBSD due to
|
||||
missing crypt.h
|
||||
|
||||
Category: Resources/res_agi
|
||||
|
||||
ASTERISK-26124: res_agi: Set audio format for EAGI audio stream
|
||||
Reported by: John Fawcett
|
||||
* [90237dca11] Sean Bright -- res_agi: Allow configuration of audio
|
||||
format of EAGI pipe
|
||||
|
||||
Category: Resources/res_pjsip_mwi
|
||||
|
||||
ASTERISK-26230: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP
|
||||
taskprocessor on startup
|
||||
Reported by: Alexei Gradinari
|
||||
* [0f6a9617eb] Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI
|
||||
subscriptions on updating contact
|
||||
* [59c9bbe696] Alexei Gradinari -- res_pjsip_mwi: don't create mwi
|
||||
subscriptions if initial unsolicited disabled
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Open Issues
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all open issues from the issue tracker that were
|
||||
referenced by changes that went into this release.
|
||||
|
||||
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
|
||||
Reported by: Etienne Allovon
|
||||
* [b07b216235] Joshua Colp -- manager: Clear the flag on the other
|
||||
channel.
|
||||
|
||||
Category: Channels/chan_pjsip
|
||||
|
||||
ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite,
|
||||
we're not trying UPDATE
|
||||
Reported by: George Joseph
|
||||
* [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'
|
||||
bridge is muted multiple times.
|
||||
Reported by: Chris Howard
|
||||
* [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
|
||||
* [0426b1d88a] Joshua Colp -- res_rtp_asterisk: Fix issues with ICE
|
||||
renegotiation.
|
||||
|
||||
Category: Resources/res_corosync
|
||||
|
||||
ASTERISK-25370: res_corosync segfaults at startup with corosync version >
|
||||
2.x
|
||||
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
|
||||
behavior difference between chan_sip and res_pjsip
|
||||
Reported by: Zach R
|
||||
* [a6e4899612] Alexei Gradinari -- res_pjsip: New endpoint option
|
||||
"notify_early_inuse_ringing"
|
||||
|
||||
Category: Resources/res_pjsip_mwi
|
||||
|
||||
ASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on
|
||||
deleting the endpoint's last contact
|
||||
Reported by: Alexei Gradinari
|
||||
* [8e749c8f51] Alexei Gradinari -- res_pjsip_mwi: unsubscribe
|
||||
unsolicited MWI on deleting endpoint last contact
|
||||
|
||||
Category: Resources/res_stasis
|
||||
|
||||
ASTERISK-27059: res_stasis: Stolen channel references are leaking
|
||||
Reported by: George Joseph
|
||||
* [edfdb4dff5] George Joseph -- res_stasis: Plug reference leak on
|
||||
stolen channels
|
||||
|
||||
Category: Third-Party/pjproject
|
||||
|
||||
ASTERISK-27097: pjproject_bundled: We don't pass options needed for
|
||||
cross-compile to pjproject configure
|
||||
Reported by: George Joseph
|
||||
* [bbe68f139d] George Joseph -- pjproject_bundled: Allow passing
|
||||
configure options to bundled
|
||||
|
||||
Improvement
|
||||
|
||||
Category: Applications/app_voicemail/IMAP
|
||||
|
||||
ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to
|
||||
specify post-polling disconnect
|
||||
Reported by: Alexei Gradinari
|
||||
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
|
||||
control
|
||||
|
||||
Category: Channels/chan_pjsip
|
||||
|
||||
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
|
||||
Reported by: Torrey Searle
|
||||
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
|
||||
|
||||
Category: Resources/res_pjsip
|
||||
|
||||
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
|
||||
Reported by: Torrey Searle
|
||||
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Commits Not Associated with an Issue
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all changes that went into this release that did not
|
||||
reference a JIRA issue.
|
||||
|
||||
+------------------------------------------------------------------------+
|
||||
| Revision | Author | Summary |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 379fe65831 | George Joseph | Fix alembic branches |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 905d18e8bf | Richard Mudgett | pjsip_distributor.c: Fix |
|
||||
| | | unidentified_requests hash functions. |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 1f59d08924 | Torrey Searle | res/res_pjsip_t38: fix incorrect |
|
||||
| | | increment of media_count |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 764d04fa87 | Richard Mudgett | res_pjsip_mwi.c: Eliminate RAII_VAR in |
|
||||
| | | contact delete observer |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| cecf6540dc | Rodrigo RamArez | cdr: fix mistake spelling of a word |
|
||||
| | Norambuena | for Unanswered. |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| b9a4ab8c8c | Richard Mudgett | chan_pjsip: Fix PJSIP_MEDIA_OFFER |
|
||||
| | | dialplan function read. |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| f1a209d5ac | Richard Mudgett | app_voicemail.c: Fix compile error |
|
||||
| | | when IMAP enabled. |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 68de35a6a0 | David M. Lee | CFLAGS for BIND8 support |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| da3312457e | Sean Bright | codecs.conf.sample: Fix max_bandwidth |
|
||||
| | | speling error |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 590ffcaf0b | Sean Bright | eventfd: Disable during cross |
|
||||
| | | compilation |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 5520b6c201 | Alexei Gradinari | CHANGES: correct version for a new |
|
||||
| | | option 'refer_blind_progress' |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| c093bf8072 | Sean Bright | res_rtp_multicast: Use consistent |
|
||||
| | | timestamps when possible |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| c10341646d | George Joseph | test_json: Fix test names with |
|
||||
| | | reserved words |
|
||||
|------------+------------------+----------------------------------------|
|
||||
| 65898c3af8 | George Joseph | unittests: Add a unit test that causes |
|
||||
| | | a SEGV and... |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
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.
|
||||
|
||||
.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(-)
|
||||
44
contrib/realtime/mssql/mssql_cdr.sql
Normal file
44
contrib/realtime/mssql/mssql_cdr.sql
Normal file
@@ -0,0 +1,44 @@
|
||||
BEGIN TRANSACTION;
|
||||
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
GO
|
||||
|
||||
-- Running upgrade -> 210693f3123d
|
||||
|
||||
CREATE TABLE cdr (
|
||||
accountcode VARCHAR(20) NULL,
|
||||
src VARCHAR(80) NULL,
|
||||
dst VARCHAR(80) NULL,
|
||||
dcontext VARCHAR(80) NULL,
|
||||
clid VARCHAR(80) NULL,
|
||||
channel VARCHAR(80) NULL,
|
||||
dstchannel VARCHAR(80) NULL,
|
||||
lastapp VARCHAR(80) NULL,
|
||||
lastdata VARCHAR(80) NULL,
|
||||
start DATETIME NULL,
|
||||
answer DATETIME NULL,
|
||||
[end] DATETIME NULL,
|
||||
duration INTEGER NULL,
|
||||
billsec INTEGER NULL,
|
||||
disposition VARCHAR(45) NULL,
|
||||
amaflags VARCHAR(45) NULL,
|
||||
userfield VARCHAR(256) NULL,
|
||||
uniqueid VARCHAR(150) NULL,
|
||||
linkedid VARCHAR(150) NULL,
|
||||
peeraccount VARCHAR(20) NULL,
|
||||
sequence INTEGER NULL
|
||||
);
|
||||
|
||||
GO
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
|
||||
|
||||
GO
|
||||
|
||||
COMMIT;
|
||||
|
||||
GO
|
||||
|
||||
1759
contrib/realtime/mssql/mssql_config.sql
Normal file
1759
contrib/realtime/mssql/mssql_config.sql
Normal file
File diff suppressed because it is too large
Load Diff
54
contrib/realtime/mssql/mssql_voicemail.sql
Normal file
54
contrib/realtime/mssql/mssql_voicemail.sql
Normal file
@@ -0,0 +1,54 @@
|
||||
BEGIN TRANSACTION;
|
||||
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
GO
|
||||
|
||||
-- Running upgrade -> a2e9769475e
|
||||
|
||||
CREATE TABLE voicemail_messages (
|
||||
dir VARCHAR(255) NOT NULL,
|
||||
msgnum INTEGER NOT NULL,
|
||||
context VARCHAR(80) NULL,
|
||||
macrocontext VARCHAR(80) NULL,
|
||||
callerid VARCHAR(80) NULL,
|
||||
origtime INTEGER NULL,
|
||||
duration INTEGER NULL,
|
||||
recording IMAGE NULL,
|
||||
flag VARCHAR(30) NULL,
|
||||
category VARCHAR(30) NULL,
|
||||
mailboxuser VARCHAR(30) NULL,
|
||||
mailboxcontext VARCHAR(30) NULL,
|
||||
msg_id VARCHAR(40) NULL
|
||||
);
|
||||
|
||||
GO
|
||||
|
||||
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||
|
||||
GO
|
||||
|
||||
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
||||
|
||||
GO
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||
|
||||
GO
|
||||
|
||||
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||
|
||||
ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE;
|
||||
|
||||
GO
|
||||
|
||||
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
|
||||
|
||||
GO
|
||||
|
||||
COMMIT;
|
||||
|
||||
GO
|
||||
|
||||
32
contrib/realtime/mysql/mysql_cdr.sql
Normal file
32
contrib/realtime/mysql/mysql_cdr.sql
Normal file
@@ -0,0 +1,32 @@
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
-- Running upgrade -> 210693f3123d
|
||||
|
||||
CREATE TABLE cdr (
|
||||
accountcode VARCHAR(20),
|
||||
src VARCHAR(80),
|
||||
dst VARCHAR(80),
|
||||
dcontext VARCHAR(80),
|
||||
clid VARCHAR(80),
|
||||
channel VARCHAR(80),
|
||||
dstchannel VARCHAR(80),
|
||||
lastapp VARCHAR(80),
|
||||
lastdata VARCHAR(80),
|
||||
start DATETIME,
|
||||
answer DATETIME,
|
||||
end DATETIME,
|
||||
duration INTEGER,
|
||||
billsec INTEGER,
|
||||
disposition VARCHAR(45),
|
||||
amaflags VARCHAR(45),
|
||||
userfield VARCHAR(256),
|
||||
uniqueid VARCHAR(150),
|
||||
linkedid VARCHAR(150),
|
||||
peeraccount VARCHAR(20),
|
||||
sequence INTEGER
|
||||
);
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
|
||||
|
||||
1070
contrib/realtime/mysql/mysql_config.sql
Normal file
1070
contrib/realtime/mysql/mysql_config.sql
Normal file
File diff suppressed because it is too large
Load Diff
34
contrib/realtime/mysql/mysql_voicemail.sql
Normal file
34
contrib/realtime/mysql/mysql_voicemail.sql
Normal file
@@ -0,0 +1,34 @@
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
-- Running upgrade -> a2e9769475e
|
||||
|
||||
CREATE TABLE voicemail_messages (
|
||||
dir VARCHAR(255) NOT NULL,
|
||||
msgnum INTEGER NOT NULL,
|
||||
context VARCHAR(80),
|
||||
macrocontext VARCHAR(80),
|
||||
callerid VARCHAR(80),
|
||||
origtime INTEGER,
|
||||
duration INTEGER,
|
||||
recording BLOB,
|
||||
flag VARCHAR(30),
|
||||
category VARCHAR(30),
|
||||
mailboxuser VARCHAR(30),
|
||||
mailboxcontext VARCHAR(30),
|
||||
msg_id VARCHAR(40)
|
||||
);
|
||||
|
||||
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||
|
||||
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||
|
||||
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||
|
||||
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
|
||||
|
||||
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
|
||||
|
||||
38
contrib/realtime/oracle/oracle_cdr.sql
Normal file
38
contrib/realtime/oracle/oracle_cdr.sql
Normal file
@@ -0,0 +1,38 @@
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR2(32 CHAR) NOT NULL
|
||||
)
|
||||
|
||||
/
|
||||
|
||||
-- Running upgrade -> 210693f3123d
|
||||
|
||||
CREATE TABLE cdr (
|
||||
accountcode VARCHAR2(20 CHAR),
|
||||
src VARCHAR2(80 CHAR),
|
||||
dst VARCHAR2(80 CHAR),
|
||||
dcontext VARCHAR2(80 CHAR),
|
||||
clid VARCHAR2(80 CHAR),
|
||||
channel VARCHAR2(80 CHAR),
|
||||
dstchannel VARCHAR2(80 CHAR),
|
||||
lastapp VARCHAR2(80 CHAR),
|
||||
lastdata VARCHAR2(80 CHAR),
|
||||
"start" DATE,
|
||||
answer DATE,
|
||||
end DATE,
|
||||
duration INTEGER,
|
||||
billsec INTEGER,
|
||||
disposition VARCHAR2(45 CHAR),
|
||||
amaflags VARCHAR2(45 CHAR),
|
||||
userfield VARCHAR2(256 CHAR),
|
||||
uniqueid VARCHAR2(150 CHAR),
|
||||
linkedid VARCHAR2(150 CHAR),
|
||||
peeraccount VARCHAR2(20 CHAR),
|
||||
sequence INTEGER
|
||||
)
|
||||
|
||||
/
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d')
|
||||
|
||||
/
|
||||
|
||||
1753
contrib/realtime/oracle/oracle_config.sql
Normal file
1753
contrib/realtime/oracle/oracle_config.sql
Normal file
File diff suppressed because it is too large
Load Diff
48
contrib/realtime/oracle/oracle_voicemail.sql
Normal file
48
contrib/realtime/oracle/oracle_voicemail.sql
Normal file
@@ -0,0 +1,48 @@
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR2(32 CHAR) NOT NULL
|
||||
)
|
||||
|
||||
/
|
||||
|
||||
-- Running upgrade -> a2e9769475e
|
||||
|
||||
CREATE TABLE voicemail_messages (
|
||||
dir VARCHAR2(255 CHAR) NOT NULL,
|
||||
msgnum INTEGER NOT NULL,
|
||||
context VARCHAR2(80 CHAR),
|
||||
macrocontext VARCHAR2(80 CHAR),
|
||||
callerid VARCHAR2(80 CHAR),
|
||||
origtime INTEGER,
|
||||
duration INTEGER,
|
||||
recording BLOB,
|
||||
flag VARCHAR2(30 CHAR),
|
||||
category VARCHAR2(30 CHAR),
|
||||
mailboxuser VARCHAR2(30 CHAR),
|
||||
mailboxcontext VARCHAR2(30 CHAR),
|
||||
msg_id VARCHAR2(40 CHAR)
|
||||
)
|
||||
|
||||
/
|
||||
|
||||
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum)
|
||||
|
||||
/
|
||||
|
||||
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir)
|
||||
|
||||
/
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e')
|
||||
|
||||
/
|
||||
|
||||
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||
|
||||
ALTER TABLE voicemail_messages MODIFY recording BLOB
|
||||
|
||||
/
|
||||
|
||||
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e'
|
||||
|
||||
/
|
||||
|
||||
36
contrib/realtime/postgresql/postgresql_cdr.sql
Normal file
36
contrib/realtime/postgresql/postgresql_cdr.sql
Normal file
@@ -0,0 +1,36 @@
|
||||
BEGIN;
|
||||
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
-- Running upgrade -> 210693f3123d
|
||||
|
||||
CREATE TABLE cdr (
|
||||
accountcode VARCHAR(20),
|
||||
src VARCHAR(80),
|
||||
dst VARCHAR(80),
|
||||
dcontext VARCHAR(80),
|
||||
clid VARCHAR(80),
|
||||
channel VARCHAR(80),
|
||||
dstchannel VARCHAR(80),
|
||||
lastapp VARCHAR(80),
|
||||
lastdata VARCHAR(80),
|
||||
start TIMESTAMP WITHOUT TIME ZONE,
|
||||
answer TIMESTAMP WITHOUT TIME ZONE,
|
||||
"end" TIMESTAMP WITHOUT TIME ZONE,
|
||||
duration INTEGER,
|
||||
billsec INTEGER,
|
||||
disposition VARCHAR(45),
|
||||
amaflags VARCHAR(45),
|
||||
userfield VARCHAR(256),
|
||||
uniqueid VARCHAR(150),
|
||||
linkedid VARCHAR(150),
|
||||
peeraccount VARCHAR(20),
|
||||
sequence INTEGER
|
||||
);
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
|
||||
|
||||
COMMIT;
|
||||
|
||||
1152
contrib/realtime/postgresql/postgresql_config.sql
Normal file
1152
contrib/realtime/postgresql/postgresql_config.sql
Normal file
File diff suppressed because it is too large
Load Diff
38
contrib/realtime/postgresql/postgresql_voicemail.sql
Normal file
38
contrib/realtime/postgresql/postgresql_voicemail.sql
Normal file
@@ -0,0 +1,38 @@
|
||||
BEGIN;
|
||||
|
||||
CREATE TABLE alembic_version (
|
||||
version_num VARCHAR(32) NOT NULL
|
||||
);
|
||||
|
||||
-- Running upgrade -> a2e9769475e
|
||||
|
||||
CREATE TABLE voicemail_messages (
|
||||
dir VARCHAR(255) NOT NULL,
|
||||
msgnum INTEGER NOT NULL,
|
||||
context VARCHAR(80),
|
||||
macrocontext VARCHAR(80),
|
||||
callerid VARCHAR(80),
|
||||
origtime INTEGER,
|
||||
duration INTEGER,
|
||||
recording BYTEA,
|
||||
flag VARCHAR(30),
|
||||
category VARCHAR(30),
|
||||
mailboxuser VARCHAR(30),
|
||||
mailboxcontext VARCHAR(30),
|
||||
msg_id VARCHAR(40)
|
||||
);
|
||||
|
||||
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
|
||||
|
||||
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
|
||||
|
||||
-- Running upgrade a2e9769475e -> 39428242f7f5
|
||||
|
||||
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
|
||||
|
||||
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
|
||||
|
||||
COMMIT;
|
||||
|
||||
Reference in New Issue
Block a user