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Author SHA1 Message Date
kharwell
56487be6d9 ChangeLog: Updated for 13.6.0 2015-10-09 16:48:48 -05:00
kharwell
68d2a14e1a Release summaries: Add summaries for 13.6.0 2015-10-09 16:48:41 -05:00
Kevin Harwell
cc0eff5651 Release summaries: Remove previous versions 2015-10-09 16:45:28 -05:00
kharwell
8cd191b885 .version: Update for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
a3777c24fd .lastclean: Update for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
68121cef21 realtime: Add database scripts for 13.6.0 2015-10-09 16:45:28 -05:00
kharwell
d72dab4f40 ChangeLog: Updated for 13.6.0-rc3 2015-10-07 13:42:02 -05:00
kharwell
9da83dbd15 Release summaries: Add summaries for 13.6.0-rc3 2015-10-07 13:41:55 -05:00
Kevin Harwell
8c60f9326c Release summaries: Remove previous versions 2015-10-07 13:41:47 -05:00
kharwell
316d47755b .version: Update for 13.6.0-rc3 2015-10-07 13:41:47 -05:00
kharwell
74a86d0a72 .lastclean: Update for 13.6.0-rc3 2015-10-07 13:41:47 -05:00
kharwell
4c39bea6f0 realtime: Add database scripts for 13.6.0-rc3 2015-10-07 13:41:46 -05:00
Matt Jordan
10e790f81a res/res_rtp_asterisk: Fix assignment after ao2 decrement
When we decide we will no longer schedule an RTCP write, we remove the
reference to the RTP instance, then assign -1 to the stored scheduler ID
in case something else comes along and wants to see if anything is scheduled.

That scheduler ID is on the RTP instance. After 60a9172d7e was merged to
fix the regression introduced by 3cf0f29310, this improper assignment on a
potentially destroyed object started getting tripped on the build agents.

Frankly, this should have been crashing a lot more often earlier. I can only
assume that the timing was changed just enough by both changes to start
actually hitting this problem.

As it is, simply moving the assignment prior to the ao2 deference is sufficient
to keep the RTP instance from being referenced when it is very, truly,
aboslutely dead.

(Note that it is still good practice to assign -1 to the scheduler ID when we
know we won't be scheduling it again, as the ao2 deref *may* not always destroy
the ao2 object.)

ASTERISK-25449

Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7
2015-10-06 20:52:21 -05:00
kharwell
c3521e9469 ChangeLog: Updated for 13.6.0-rc2 2015-10-06 13:40:31 -05:00
kharwell
a44f6aa046 Release summaries: Add summaries for 13.6.0-rc2 2015-10-06 13:40:21 -05:00
Kevin Harwell
dd74af7e46 Release summaries: Remove previous versions 2015-10-06 13:40:13 -05:00
kharwell
a11a78ca34 .version: Update for 13.6.0-rc2 2015-10-06 13:40:12 -05:00
kharwell
570329ec8a .lastclean: Update for 13.6.0-rc2 2015-10-06 13:40:12 -05:00
kharwell
51c9ff47f6 realtime: Add database scripts for 13.6.0-rc2 2015-10-06 13:40:12 -05:00
Joshua Colp
5c65449e04 Merge "res_rtp_asterisk: Move "Set role" warning to be debug." into 13.6 2015-10-06 09:02:06 -05:00
Matt Jordan
754daeca0a Fix improper usage of scheduler exposed by 5c713fdf18
When 5c713fdf18 was merged, it allowed for scheduled items to have an ID of
'0' returned. While this was valid per the documentation for the API, it was
apparently never returned previously. As a result, several users of the
scheduler API viewed the result as being invalid, causing them to reschedule
already scheduled items or otherwise fail in interesting ways.

This patch corrects the users such that they view '0' as valid, and a returned
ID of -1 as being invalid.

Note that the failing HEP RTCP tests now pass with this patch. These tests
failed due to a duplicate scheduling of the RTCP transmissions.

ASTERISK-25449 #close

Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39
2015-10-06 07:40:22 -05:00
Joshua Colp
9913d47697 res_rtp_asterisk: Move "Set role" warning to be debug.
In practice the set_role API callback can be invoked even
when no ICE is present on an RTP instance. This can occur
if ICE has not been enabled on it.

ASTERISK-25438 #close

Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69
(cherry picked from commit 56ed7b9dd5)
2015-10-06 05:55:10 -05:00
kharwell
a0fb436eda ChangeLog: Updated for 13.6.0-rc1 2015-09-29 16:15:34 -05:00
kharwell
bba1c4066b Release summaries: Add summaries for 13.6.0-rc1 2015-09-29 16:12:02 -05:00
kharwell
82c4aecdbb .version: Update for 13.6.0-rc1 2015-09-29 16:08:20 -05:00
kharwell
bc18db7388 .lastclean: Update for 13.6.0-rc1 2015-09-29 16:08:20 -05:00
kharwell
b9c53f95e3 realtime: Add database scripts for 13.6.0-rc1 2015-09-29 16:08:20 -05:00
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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-13.6.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.6.0</h3><h3 align="center">Date: 2015-10-09</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#open_issues">Open Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><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-13.5.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">40 Richard Mudgett <rmudgett@digium.com><br/>18 Kevin Harwell <kharwell@lunkwill><br/>16 Joshua Colp <jcolp@digium.com><br/>15 Mark Michelson <mmichelson@digium.com><br/>13 Matt Jordan <mjordan@digium.com><br/>9 Scott Griepentrog <scott@griepentrog.com><br/>8 Kevin Harwell <kharwell@lunkwill.digium.internal><br/>3 Scott Emidy <jemidy@digium.com><br/>3 David M. Lee <dlee@respoke.io><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Alexander Anikin <may213@yandex.ru><br/>2 Jonathan Rose <jrose@digium.com><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>1 Mark Duncan <mark@syon.co.jp><br/>1 Benjamin Ford <bford@digium.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/></td><td width="33%">1 Elazar Broad<br/></td><td width="33%">13 Matt Jordan <mjordan@digium.com><br/>11 Joshua Colp <jcolp@digium.com><br/>11 Richard Mudgett <rmudgett@digium.com><br/>10 Scott Griepentrog <sgriepentrog@digium.com><br/>8 Mark Michelson<br/>8 Mark Michelson <mmichelson@digium.com><br/>7 John Hardin<br/>5 Kevin Harwell <kharwell@digium.com><br/>4 Scott Griepentrog<br/>3 Richard Mudgett<br/>2 Kevin Harwell<br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Alexandr Dranchuk <alex.dranchuk@gmail.com><br/>2 Dmitriy Serov <serov.d.p@gmail.com><br/>2 Stefan Engström <stefanen@kth.se><br/>1 Oleg Kozlov <olkeep@gmail.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Etienne Lessard<br/>1 Lorne Gaetz<br/>1 Chet Stevens <cwstevens@interact.ccsd.net><br/>1 Rodrigo Ramirez Norambuena <decipher.hk@gmail.com><br/>1 Ashley Sanders<br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Chet Stevens<br/>1 Kevin Scott Adams <ksatllc@att.net><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Sean Pimental<br/>1 Jonathan Rose <jrose@digium.com><br/>1 Etienne Lessard <elessard@avencall.com><br/>1 Alexandr Dranchuk<br/>1 yaron nahum <nachum.yaron@gmail.com><br/>1 Elazar Broad<br/>1 yaron nahum<br/>1 Guido Falsi<br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>New Feature</h3><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25252">ASTERISK-25252</a>: ARI: Add the ability to manipulate log channels<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df9ce3636695781be6ab2479f90766a56747dbd7">[df9ce36366]</a> Scott Emidy -- ARI: Retrieve existing log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e9f1bc08cbda7759707c30b8883b266555d0fefc">[e9f1bc08cb]</a> Scott Emidy -- ARI: Creating log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78364132ce94d9ded24ae6e6ab44b97d256b506d">[78364132ce]</a> Scott Emidy -- ARI: Deleting log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ae762634c317fbcbd98a8c34d2474f7d4b654ed">[1ae762634c]</a> Benjamin Ford -- ARI: Rotate log channels.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25377">ASTERISK-25377</a>: res_pjsip: Change default "From user" from UUID to something more palatable<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac62928d6b7333a0d502be2eba99c238549ae1a3">[ac62928d6b]</a> Mark Michelson -- res_pjsip: Change default from user value.</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25227">ASTERISK-25227</a>: No audio at in-band announcements in ooh323 channel<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71408df2b82c932329d250a6077475e4f51a2b0d">[71408df2b8]</a> Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE with inband info available</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25322">ASTERISK-25322</a>: Crash occurs when using MixMonitor with t() or r() options.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3a56bee83c01dbb538620a11947d420b17cf458">[b3a56bee83]</a> Richard Mudgett -- audiohook.c: Fix MixMonitor crash when using the r() or t() options.</li>
</ul><br><h4>Category: Applications/app_page</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25384">ASTERISK-25384</a>: Regular Asterisk crashes when using Page application. "user_data is NULL"<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f15cd93f0b2cb622d54061515b815e3ebbe76b1">[5f15cd93f0]</a> Richard Mudgett -- app_page.c: Fix crash when forwarding with a predial handler.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25399">ASTERISK-25399</a>: app_queue: AgentComplete event has wrong reason<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4fb95bbc4e4928dd3403a20d401c285a568f0d09">[4fb95bbc4e]</a> Kevin Harwell -- app_queue: AgentComplete event has wrong reason</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25185">ASTERISK-25185</a>: Segfault in app_queue on transfer scenarios<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6409e7b11a2310196a9978b30a6b79e2760be592">[6409e7b11a]</a> Kevin Harwell -- app_queue: Crash when transferring</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25215">ASTERISK-25215</a>: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember<br/>Reported by: Lorne Gaetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5f5b9f384eef0389a7e35d40c91d3586869a125">[e5f5b9f384]</a> Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.</li>
</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25410">ASTERISK-25410</a>: app_record: RECORDED_FILE variable not being populated<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeddee39fb492ea5ec873bdf02ea3858c5282601">[aeddee39fb]</a> Kevin Harwell -- app_record: RECORDED_FILE variable not being populated</li>
</ul><br><h4>Category: Bridges/bridge_holding</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25271">ASTERISK-25271</a>: Parking & blind transfer: Transferer channel not hung up if no MOH<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8458b8d441c2f4143ff135163ff3da4f88fe14c8">[8458b8d441]</a> Jonathan Rose -- holding_bridge: ensure moh participants get frames</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25315">ASTERISK-25315</a>: DAHDI channels send shortened duration DTMF tones.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=256bc52b6684141b100c5018dbf0ad3ce6111585">[256bc52b66]</a> Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=800e0ea48de88c016e1f477edf7db7b9aadc4b54">[800e0ea48d]</a> Richard Mudgett -- chan_dahdi.c: Lock private struct for ast_write().</li>
</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25309">ASTERISK-25309</a>: [patch] iLBC 20 advertised<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f68c995bc97c9b6cb4887043b344087d82aeef10">[f68c995bc9]</a> Alexander Traud -- chan_sip: Fix negotiation of iLBC 30.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25346">ASTERISK-25346</a>: chan_sip: Overwriting answered elsewhere hangup cause on call pickup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c01111223f9dbd383a4dd1cf786b63eff214f238">[c01111223f]</a> Joshua Colp -- chan_sip: Allow call pickup to set the hangup cause.</li>
</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25396">ASTERISK-25396</a>: chan_sip: Extremely long callerid name causes invalid SIP<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b59c4d82b58a7a10e1791f9ed5759af8ac637df2">[b59c4d82b5]</a> Walter Doekes -- chan_sip: Fix From header truncation for extremely long CALLERID(name).</li>
</ul><br><h4>Category: Channels/chan_sip/Security Framework</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25320">ASTERISK-25320</a>: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25af2d71c863062868b8bda6cf83515a1935d27e">[25af2d71c8]</a> Kevin Harwell -- chan_sip.c: wrong peer searched in sip_report_security_event</li>
</ul><br><h4>Category: Channels/chan_skinny</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25353">ASTERISK-25353</a>: [patch] Transcoding while different in Frame size = Frames lost<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b88c54fa4bd537bde46519abb95e30a5f96673ac">[b88c54fa4b]</a> Alexander Traud -- translate: Fix transcoding while different in frame size.</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25341">ASTERISK-25341</a>: bridge: Hangups may get lost when executing actions<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c2dab1e888e59cb429ed61219815bd00eee66c0">[6c2dab1e88]</a> Joshua Colp -- bridge: Kick channel from bridge if hung up during action.</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25449">ASTERISK-25449</a>: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=10e790f81ad3be322193cf948e0334f9af09b00f">[10e790f81a]</a> Matt Jordan -- res/res_rtp_asterisk: Fix assignment after ao2 decrement</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=754daeca0a48e5c9365cc7fa4c5a3da6c61ae7f6">[754daeca0a]</a> Matt Jordan -- Fix improper usage of scheduler exposed by 5c713fdf18f</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25418">ASTERISK-25418</a>: On-hold channels redirected out of a bridge appear to still be on hold<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=629458d34930e5aca56f749bc05562baf95d13f7">[629458d349]</a> Mark Michelson -- Do not swallow frames on channels leaving bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25355">ASTERISK-25355</a>: sched: ast_sched_del may return prematurely due to spurious wakeup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85e1cb51b21c2d194647e16b81b5a1344d2ff911">[85e1cb51b2]</a> Joshua Colp -- sched: ast_sched_del may return prematurely due to spurious wakeup</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25305">ASTERISK-25305</a>: Dynamic logger channels can be added multiple times<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f050fa76eb8535a2b8a3b047527a42ea369d8792">[f050fa76eb]</a> Mark Michelson -- logger: Prevent duplicate dynamic channels from being added.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25407">ASTERISK-25407</a>: Asterisk fails to log to multiple syslog destinations<br/>Reported by: Elazar Broad<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec514ad64dbc0014525008977c8c74c2856c9d3a">[ec514ad64d]</a> Elazar Broad -- core/logging: Fix logging to more than one syslog channel</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25394">ASTERISK-25394</a>: pbx: Incorrect device and presence state when changing hint details<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2bd27d12223fe33b58c453965ed5c6ed3af7c4f5">[2bd27d1222]</a> Joshua Colp -- pbx: Update device and presence state when changing a hint extension.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25367">ASTERISK-25367</a>: pbx: Long pattern match hints may cause "core show hints" to crash<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc1363209e7a5b6c67bf96c593e3beb0884c1fb0">[cc1363209e]</a> Joshua Colp -- pbx: Fix crash when issuing "core show hints" with long pattern match.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25362">ASTERISK-25362</a>: Deadlock due to presence state callback<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03fe79f29eae42be24589b323a5ef3fa9259158d">[03fe79f29e]</a> Mark Michelson -- Fix deadlock on presence state changes.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25325">ASTERISK-25325</a>: ARI PUT reload chan_sip HTTP response 404<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=865377fc38134234f17def6634c47a989cf0e77a">[865377fc38]</a> Rodrigo Ramírez Norambuena -- chan_sip.c: Validation on module reload</li>
</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25312">ASTERISK-25312</a>: res_http_websocket: Terminate connection on fatal cases<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4e9416138339274176cb87b26db905723e553ba">[b4e9416138]</a> Joshua Colp -- res_http_websocket: Forcefully terminate on write errors.</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25369">ASTERISK-25369</a>: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fbf720db91ae8942e9c2ba092179ab2352d44b06">[fbf720db91]</a> Jonathan Rose -- ParkAndAnnounce: Add variable inheritance</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25295">ASTERISK-25295</a>: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5469caa9ddf002d2e75b5fe5dec0c4dbebea1d1e">[5469caa9dd]</a> Joshua Colp -- res_pjsip: Use hash for contact object identity instead of Contact URI.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a676ba2aad5525926ae31b8317b95ae52cbbabbb">[a676ba2aad]</a> Joshua Colp -- taskprocessor: Fix race condition between unreferencing and finding.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25381">ASTERISK-25381</a>: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3e6debdb95a5895894ed2b58b600fcdf17927b9">[c3e6debdb9]</a> Matt Jordan -- res/res_pjsip: Purge contacts when an AoR is deleted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25339">ASTERISK-25339</a>: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc6fe07f5c114bdeaef4a3b83a11faaa9d1046eb">[bc6fe07f5c]</a> Matt Jordan -- res_pjsip/pjsip_configuration: Disregard empty auth values</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25304">ASTERISK-25304</a>: res_pjsip: XML sanitization may write past buffer<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8521a86367ac6090210a89878c8fee6d19c43642">[8521a86367]</a> Joshua Colp -- res_pjsip: Ensure sanitized XML is NULL terminated.</li>
</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25387">ASTERISK-25387</a>: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1dd0e220bf98ca93b825d7b5af4160f7718eab38">[1dd0e220bf]</a> Matt Jordan -- res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25306">ASTERISK-25306</a>: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c126afe18f9073f3ee74e45f574da421131b9fa2">[c126afe18f]</a> Richard Mudgett -- res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e25569ef95c6de6e9267df4673bd1d774b82a000">[e25569ef95]</a> Mark Michelson -- res_pjsip_pubsub: More accurately persist packet.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25356">ASTERISK-25356</a>: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b1561f4c854c37691bd24227b8f722d1dac4291">[1b1561f4c8]</a> Joshua Colp -- res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25297">ASTERISK-25297</a>: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13eb491e35ae6a99164dec6a62d7f05784c75c11">[13eb491e35]</a> Richard Mudgett -- res_pjsip_session.c: Fix crashes seen when call cancelled.</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25438">ASTERISK-25438</a>: res_rtp_asterisk: ICE role message even when ICE is not enabled<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9913d47697716040917b49c0c60ae2d98493d516">[9913d47697]</a> Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25318">ASTERISK-25318</a>: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2c73190825bf4c9cedb1031327199767a4a3ca8">[c2c7319082]</a> Joshua Colp -- res_pjsip_session: Don't invoke session supplements twice for BYE requests.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25292">ASTERISK-25292</a>: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=10ba72a9279c591e800ad2656d367b881f73203d">[10ba72a927]</a> Mark Michelson -- Add a test event for inband ringing.</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25310">ASTERISK-25310</a>: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ed9c9a280c08a17fce602c15d2b01de199ca736">[4ed9c9a280]</a> Guido Falsi -- Core/General: Add #ifdef needed on FreeBSD.</li>
</ul><br><h4>Category: Resources/res_ari_applications</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</li>
</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</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: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25299">ASTERISK-25299</a>: RTP port leaks with incoming OOH323 calls<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=480c443e2691272a7227e0949244e80e53bc31b2">[480c443e26]</a> Alexander Anikin -- chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25323">ASTERISK-25323</a>: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c94f46080f60435fffd197d14441ccf9d963521b">[c94f46080f]</a> Scott Griepentrog -- CHAOS: avoid crash if string create fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cc59533b903b3d55e8b388f28385287e712ae62">[4cc59533b9]</a> Richard Mudgett -- CHAOS: res_pjsip_diversion avoid crash if allocation fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb6b5c684b8772ba008339a417725a208f72409e">[fb6b5c684b]</a> Scott Griepentrog -- PJSIP: avoid crash when getting rtp peer</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f72f9ceefca52002c45f5910219dbcb0f9437a79">[f72f9ceefc]</a> Scott Griepentrog -- pjsip: avoid possible crash req_caps allocation failure</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6862c2a167f4ed2cb8511bb1ae94a13582afa25b">[6862c2a167]</a> Scott Griepentrog -- Chaos: handle failed allocation in get_media_encryption_type</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1cd6366588c66dce5be66541ceb7f828fde3773">[f1cd636658]</a> Scott Griepentrog -- Chaos: make hangup NULL tolerant</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab373f2ceffcad3a497663027199f4f4a81f644b">[ab373f2cef]</a> Scott Griepentrog -- CHAOS: prevent sorcery object with null id</li>
</ul><br><h4>Category: Resources/res_hep_rtcp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25352">ASTERISK-25352</a>: res_hep_rtcp correlation_id is different then res_hep<br/>Reported by: Kevin Scott Adams<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78d0b9d97ecf034468e252440217dd4bc371ef71">[78d0b9d97e]</a> Matt Jordan -- channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24602">ASTERISK-24602</a>: Unable to call WebRTC client via wss on chan_pjsip<br/>Reported by: Oleg Kozlov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d32e516c7cd48979db092a82b97a7ac4a743f526">[d32e516c7c]</a> Martin Tomec -- res/pjsip: Mark WSS transport as secure</li>
</ul><br><h3>Improvement</h3><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=cc0eff5651e792d9f97a433fbbac99992c20cb46">cc0eff5651</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8cd191b885c5410faa11e28b52d37fea9d5c5197">8cd191b885</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3777c24fd93fe757ce8c68daf9c2749938eea8f">a3777c24fd</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68121cef21b3753a28248b493d211a6d3176e67a">68121cef21</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d72dab4f402c06b304e339f4e8fd801b502cb003">d72dab4f40</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9da83dbd1524aa1c74907066ec6480e9d56651f9">9da83dbd15</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c60f9326ce609061316f4bc3b1522caa7282482">8c60f9326c</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=316d47755b0d6239cb5bbd674d7596046ad4cddd">316d47755b</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74a86d0a72c9858dc965e8e2b59ae4a1e0294033">74a86d0a72</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c39bea6f00560dd6bc52d9d793e1ae7ebf130b7">4c39bea6f0</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc3</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3521e9469f95b962e52edbbb6e0cdef6d3ceab4">c3521e9469</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a44f6aa046c988f3c1ce9defba5038f83255a4c4">a44f6aa046</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd74af7e460083df8ca774db2793791ff6413c6e">dd74af7e46</a></td><td>Kevin Harwell</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a11a78ca34fae7ef2069d0a9ae6a5cf1fd2856d7">a11a78ca34</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=570329ec8a05c27675721734e2da3c42c707f0f7">570329ec8a</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51c9ff47f6bab1aed30f95a8601e0ddf253b657e">51c9ff47f6</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0fb436eda914dca26e96d304d3c9daca2be54de">a0fb436eda</a></td><td>Kevin Harwell</td><td>ChangeLog: Updated for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bba1c4066be4d752f106266a7c084ba2340c2777">bba1c4066b</a></td><td>Kevin Harwell</td><td>Release summaries: Add summaries for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82c4aecdbb4466843a41e0591ed0e7c16c1902a6">82c4aecdbb</a></td><td>Kevin Harwell</td><td>.version: Update for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc18db73884513e5636fc4af8d1b8bf2ba355c3a">bc18db7388</a></td><td>Kevin Harwell</td><td>.lastclean: Update for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9c53f95e38e7e9e47c9843bcbb0e8836cafed1b">b9c53f95e3</a></td><td>Kevin Harwell</td><td>realtime: Add database scripts for 13.6.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d30939b6e84f18eb1fa3eb9819951fe8a1c764f4">d30939b6e8</a></td><td>Kevin Harwell</td><td>ARI: Changed version from 1.8.0 to 1.9.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f19c9baded56a5adb419a7bcb1ac00fbe09f404">5f19c9bade</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Fix user sort compare function.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a8576403968bd1f49c8bef67735c04d05fb6983">3a85764039</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Optimize conf_alloc() object init.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbeda190c3f05e95a82c7d9609c66dcf3ce35bd3">bbeda190c3</a></td><td>Richard Mudgett</td><td>app_dial.c: Remove some no-op code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe5077b1f8caed2df419e1bd7b872657b7def726">fe5077b1f8</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Eliminate race during initial NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c713fdf18ffa934e0cac8ddb29e4ad95a68200b">5c713fdf18</a></td><td>Mark Michelson</td><td>scheduler: Use queue for allocating sched IDs.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e75aff53e6ee68833595db101e43329adf9a4459">e75aff53e6</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d91d01df180a712485aa60d14fda2aa9e5063d2">4d91d01df1</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add some notification comments.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f36a9d122171781dd8a99d32792a0c19103b1f15">f36a9d1221</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94582f8fabb926d197ce0f3a01208b385975ec09">94582f8fab</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix off-nominal memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b3ed52239b24546b1ee12156dadccb70db7403e">8b3ed52239</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix one byte buffer overrun error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4329bd1e4c059e714122465901ea2c46dd924b71">4329bd1e4c</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a456a20ecf835bcf70ebc2a279e230df402bec08">a456a20ecf</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add missing error return in load_module().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f58f4c6e2762e5ad7eccf7065e63b345f4cda7f6">f58f4c6e27</a></td><td>Richard Mudgett</td><td>res_pjsip/location.c: Use the builtin ao2_callback() match function instead.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4eedd9ef9d7c000cd8d67cbeb1789ac6d71860aa">4eedd9ef9d</a></td><td>Matt Jordan</td><td>main/config_options: Check for existance of internal object before derefing</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=695f26cbb759f8ecc18c6b1d6cf84b3105b2f007">695f26cbb7</a></td><td>David M. Lee</td><td>res_rtp_asterisk: Add more ICE debugging</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61c6c6aa6c60636207567a49c6320946c1840e99">61c6c6aa6c</a></td><td>David M. Lee</td><td>Fix when remote candidates exceed PJ_ICE_MAX_CAND</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad9cb6c2ce6dbe9c985c6891daf53cc4160e3a13">ad9cb6c2ce</a></td><td>Mark Michelson</td><td>res_pjsip: Fix contact refleak on stateful responses.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c4d0c3506374b89502bd6c1bda89c3f241b6708">7c4d0c3506</a></td><td>Joshua Colp</td><td>res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0582776f7fe55e72acda586fb32185ad7879aeab">0582776f7f</a></td><td>Richard Mudgett</td><td>ari/ari_websockets.c: Fix ast_debug parameter type mismatch.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77518d54344945d7d3bfc1ebfe61d97704fa5dfa">77518d5434</a></td><td>Richard Mudgett</td><td>res_http_websocket.c: Fix some off nominal path cleanup.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c61547fee6fdc1e132d359311da48e87d98d25b1">c61547fee6</a></td><td>Richard Mudgett</td><td>res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd867cd0787b124984caf2604478212651ea4c03">bd867cd078</a></td><td>Richard Mudgett</td><td>app_queue.c: Extract some functions for simpler code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ded51e3d77a13eed059a85a083d0ab0324a77db7">ded51e3d77</a></td><td>Richard Mudgett</td><td>app_queue.c: Fix error checking in QUEUE_MEMBER() read.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b719f56c72c9cc66879eeef11de2ef4498cba648">b719f56c72</a></td><td>Mark Michelson</td><td>res_pjsip_sdp_rtp: Restore removed NULL check.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cea5dc7b8afd0e8cbde4c5d253bac3219125b168">cea5dc7b8a</a></td><td>Richard Mudgett</td><td>audiohook.c: Simplify variable usage in audiohook_read_frame_both().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e18c300550453df06507e32c4bc78ef91d369f27">e18c300550</a></td><td>Joshua Colp</td><td>res_http_websocket: When shutting down a session don't close closed socket</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e194047acb3ba846356041c1a6222caefc65a2e">8e194047ac</a></td><td>Matt Jordan</td><td>res/res_format_attr_silk: Expose format attributes to other modules</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0f451c35ed56b08353c4c3150bf847867f74fe7">a0f451c35e</a></td><td>Matt Jordan</td><td>main/format: Add an API call for retrieving format attributes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26f0559a94519b735396ed4ce90e23c1d9d5b332">26f0559a94</a></td><td>David M. Lee</td><td>Replace htobe64 with htonll</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27dc2094e98b59b8a50b059ddd6048285a42e6b9">27dc2094e9</a></td><td>Mark Michelson</td><td>res_http_websocket: Debug write lengths.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39cc28f6ea2140ad6d561fd4c9e9a66f065cecee">39cc28f6ea</a></td><td>Mark Michelson</td><td>res_http_websocket: Avoid passing strlen() to ast_websocket_write().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1519eb44a796911c7c438bbe4e31bb89be244387">1519eb44a7</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Must protect mime_types_len with mime_types_lock.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a93b7a927c5975c0a8889dc66868f81e4eef8aa3">a93b7a927c</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=741fa0d26d38f7a9ed98c595ee1bb6b6ce8a9923">741fa0d26d</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fixup some whitespace.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b21fd9a38bcd89402249440c1670ce48781f30">89b21fd9a3</a></td><td>Richard Mudgett</td><td>rtp_engine.h: No sense allowing payload types larger than RFC allows.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7427c7f13b1d0c9095e83a3ea38394f521d3a75e">7427c7f13b</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Minor tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e20f435b6093c2c72e678b9fad1ed037c3191b88">e20f435b60</a></td><td>Richard Mudgett</td><td>rtp_engine.h: Misc comment fixes.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc5d7f9c37fe8a4ff5744ab8620898ccae6a7d2a">bc5d7f9c37</a></td><td>Richard Mudgett</td><td>chan_sip.c: Tweak glue-&gt;update_peer() parameter nil value.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=48698a5e21d7307f61b5fb2bd39fd593bc1423ca">48698a5e21</a></td><td>Mark Michelson</td><td>res_http_websocket: Properly encode 64 bit payload</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>asterisk-13.5.0-summary.html | 441 --
asterisk-13.5.0-summary.txt | 1141 ------
b/.version | 2
b/CHANGES | 35
b/ChangeLog | 1884 +++++++++-
b/Makefile | 1
b/UPGRADE.txt | 6
b/addons/chan_ooh323.c | 1
b/addons/ooh323c/src/ooq931.c | 6
b/apps/app_dial.c | 156
b/apps/app_page.c | 28
b/apps/app_queue.c | 412 +-
b/apps/app_record.c | 3
b/bridges/bridge_holding.c | 6
b/channels/chan_dahdi.c | 55
b/channels/chan_pjsip.c | 18
b/channels/chan_sip.c | 78
b/channels/chan_skinny.c | 26
b/channels/pjsip/dialplan_functions.c | 5
b/channels/sip/include/security_events.h | 3
b/channels/sip/security_events.c | 5
b/codecs/codec_gsm.c | 29
b/codecs/codec_ilbc.c | 28
b/codecs/codec_lpc10.c | 41
b/codecs/codec_speex.c | 60
b/configure | 63
b/configure.ac | 6
b/contrib/ast-db-manage/config/versions/154177371065_add_default_from_user.py | 22
b/contrib/realtime/mssql/mssql_config.sql | 8
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 8
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/astversion | 536 ++
b/contrib/scripts/install_prereq | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format.h | 23
b/include/asterisk/logger.h | 44
b/include/asterisk/res_pjsip.h | 40
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/rtp_engine.h | 18
b/include/asterisk/stasis_app.h | 15
b/main/audiohook.c | 43
b/main/bridge_channel.c | 7
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/dial.c | 25
b/main/endpoints.c | 3
b/main/format.c | 11
b/main/logger.c | 272 +
b/main/pbx.c | 146
b/main/rtp_engine.c | 153
b/main/sched.c | 143
b/main/sorcery.c | 4
b/main/stasis_endpoints.c | 78
b/main/taskprocessor.c | 12
b/main/translate.c | 55
b/main/utils.c | 4
b/res/ari/ari_model_validators.c | 445 ++
b/res/ari/ari_model_validators.h | 118
b/res/ari/ari_websockets.c | 9
b/res/ari/config.c | 72
b/res/ari/resource_asterisk.c | 127
b/res/ari/resource_asterisk.h | 63
b/res/ari/resource_events.c | 24
b/res/ari/resource_events.h | 2
b/res/parking/parking_applications.c | 65
b/res/res_ari.c | 19
b/res/res_ari_asterisk.c | 300 +
b/res/res_ari_events.c | 6
b/res/res_config_sqlite.c | 8
b/res/res_format_attr_silk.c | 24
b/res/res_http_websocket.c | 80
b/res/res_pjsip.c | 131
b/res/res_pjsip/config_global.c | 18
b/res/res_pjsip/location.c | 51
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/presence_xml.c | 31
b/res/res_pjsip_diversion.c | 4
b/res/res_pjsip_multihomed.c | 23
b/res/res_pjsip_nat.c | 2
b/res/res_pjsip_pidf_digium_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 80
b/res/res_pjsip_sdp_rtp.c | 70
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_t38.c | 15
b/res/res_pjsip_transport_websocket.c | 5
b/res/res_rtp_asterisk.c | 79
b/res/res_stasis.c | 45
b/res/res_stasis_device_state.c | 54
b/res/stasis/app.c | 335 +
b/res/stasis/app.h | 15
b/res/stasis/messaging.c | 44
b/rest-api-templates/ari_model_validators.c.mustache | 2
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 130
b/rest-api/api-docs/bridges.json | 2
b/rest-api/api-docs/channels.json | 2
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 109
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/tests/test_core_format.c | 57
106 files changed, 6435 insertions(+), 2535 deletions(-)</pre><br></html>

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Release Summary
asterisk-13.6.0
Date: 2015-10-09
<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-13.5.0.
----------------------------------------------------------------------
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
40 Richard Mudgett 1 Elazar Broad 13 Matt Jordan
18 Kevin Harwell 11 Joshua Colp
16 Joshua Colp 11 Richard Mudgett
15 Mark Michelson 10 Scott Griepentrog
13 Matt Jordan 8 Mark Michelson
9 Scott Griepentrog 8 Mark Michelson
8 Kevin Harwell 7 John Hardin
3 Scott Emidy 5 Kevin Harwell
3 David M. Lee 4 Scott Griepentrog
2 Alexander Traud 3 Richard Mudgett
2 Alexander Anikin 2 Kevin Harwell
2 Jonathan Rose 2 Alexander Traud
1 Martin Tomec 2 Alexandr Dranchuk
1 Elazar Broad 2 Dmitriy Serov
1 Walter Doekes 2 Stefan EngstrAP:m
1 Rodrigo RamArez Norambuena 1 Oleg Kozlov
1 Mark Duncan 1 Walter Doekes
1 Benjamin Ford 1 Etienne Lessard
1 Guido Falsi 1 Lorne Gaetz
1 Chet Stevens
1 Rodrigo Ramirez Norambuena
1 Ashley Sanders
1 Guido Falsi
1 Lorne Gaetz
1 Chet Stevens
1 Kevin Scott Adams
1 Elazar Broad
1 Sean Pimental
1 Jonathan Rose
1 Etienne Lessard
1 Alexandr Dranchuk
1 yaron nahum
1 Elazar Broad
1 yaron nahum
1 Guido Falsi
----------------------------------------------------------------------
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.
New Feature
Category: Resources/res_ari
ASTERISK-25252: ARI: Add the ability to manipulate log channels
Reported by: Matt Jordan
* [df9ce36366] Scott Emidy -- ARI: Retrieve existing log channels
* [e9f1bc08cb] Scott Emidy -- ARI: Creating log channels
* [78364132ce] Scott Emidy -- ARI: Deleting log channels
* [1ae762634c] Benjamin Ford -- ARI: Rotate log channels.
Category: Resources/res_pjsip
ASTERISK-25377: res_pjsip: Change default "From user" from UUID to
something more palatable
Reported by: Mark Michelson
* [ac62928d6b] Mark Michelson -- res_pjsip: Change default from user
value.
Bug
Category: Addons/chan_ooh323
ASTERISK-25227: No audio at in-band announcements in ooh323 channel
Reported by: Alexandr Dranchuk
* [71408df2b8] Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE
with inband info available
Category: Applications/app_dial
ASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard Mudgett
* [6b1e7583c1] Richard Mudgett -- app_queue.c: Force COLP update if
outgoing channel name changed.
* [6bf304bf25] Richard Mudgett -- app_queue.c: Factor out a connected
line update routine.
* [e36b5f1e8e] Richard Mudgett -- app_dial.c: Make 'A' option pass COLP
updates.
* [747bfac895] Richard Mudgett -- app_dial.c: Force COLP update if
outgoing channel name changed.
* [14481d9aa0] Richard Mudgett -- app_dial.c: Factor out a connected
line update routine.
Category: Applications/app_mixmonitor
ASTERISK-25322: Crash occurs when using MixMonitor with t() or r()
options.
Reported by: Richard Mudgett
* [b3a56bee83] Richard Mudgett -- audiohook.c: Fix MixMonitor crash when
using the r() or t() options.
Category: Applications/app_page
ASTERISK-25384: Regular Asterisk crashes when using Page application.
"user_data is NULL"
Reported by: Chet Stevens
* [5f15cd93f0] Richard Mudgett -- app_page.c: Fix crash when forwarding
with a predial handler.
Category: Applications/app_queue
ASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard Mudgett
* [6b1e7583c1] Richard Mudgett -- app_queue.c: Force COLP update if
outgoing channel name changed.
* [6bf304bf25] Richard Mudgett -- app_queue.c: Factor out a connected
line update routine.
* [e36b5f1e8e] Richard Mudgett -- app_dial.c: Make 'A' option pass COLP
updates.
* [747bfac895] Richard Mudgett -- app_dial.c: Force COLP update if
outgoing channel name changed.
* [14481d9aa0] Richard Mudgett -- app_dial.c: Factor out a connected
line update routine.
ASTERISK-25399: app_queue: AgentComplete event has wrong reason
Reported by: Kevin Harwell
* [4fb95bbc4e] Kevin Harwell -- app_queue: AgentComplete event has wrong
reason
ASTERISK-25185: Segfault in app_queue on transfer scenarios
Reported by: Etienne Lessard
* [6409e7b11a] Kevin Harwell -- app_queue: Crash when transferring
ASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and
using PauseQueueMember
Reported by: Lorne Gaetz
* [e5f5b9f384] Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER
'paused' and 'ringinuse'.
Category: Applications/app_record
ASTERISK-25410: app_record: RECORDED_FILE variable not being populated
Reported by: Kevin Harwell
* [aeddee39fb] Kevin Harwell -- app_record: RECORDED_FILE variable not
being populated
Category: Bridges/bridge_holding
ASTERISK-25271: Parking & blind transfer: Transferer channel not hung up
if no MOH
Reported by: Kevin Harwell
* [8458b8d441] Jonathan Rose -- holding_bridge: ensure moh participants
get frames
Category: Channels/chan_dahdi
ASTERISK-25315: DAHDI channels send shortened duration DTMF tones.
Reported by: Richard Mudgett
* [256bc52b66] Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write
buffer after starting DTMF.
* [800e0ea48d] Richard Mudgett -- chan_dahdi.c: Lock private struct for
ast_write().
Category: Channels/chan_sip/CodecHandling
ASTERISK-25309: [patch] iLBC 20 advertised
Reported by: Alexander Traud
* [f68c995bc9] Alexander Traud -- chan_sip: Fix negotiation of iLBC 30.
Category: Channels/chan_sip/General
ASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on
call pickup
Reported by: Joshua Colp
* [c01111223f] Joshua Colp -- chan_sip: Allow call pickup to set the
hangup cause.
Category: Channels/chan_sip/Interoperability
ASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP
Reported by: Walter Doekes
* [b59c4d82b5] Walter Doekes -- chan_sip: Fix From header truncation for
extremely long CALLERID(name).
Category: Channels/chan_sip/Security Framework
ASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong
or non existent peer on invite
Reported by: Kevin Harwell
* [25af2d71c8] Kevin Harwell -- chan_sip.c: wrong peer searched in
sip_report_security_event
Category: Channels/chan_skinny
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Channels/chan_unistim
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Codecs/General
ASTERISK-25353: [patch] Transcoding while different in Frame size = Frames
lost
Reported by: Alexander Traud
* [b88c54fa4b] Alexander Traud -- translate: Fix transcoding while
different in frame size.
Category: Core/Bridging
ASTERISK-25341: bridge: Hangups may get lost when executing actions
Reported by: Joshua Colp
* [6c2dab1e88] Joshua Colp -- bridge: Kick channel from bridge if hung
up during action.
Category: Core/BuildSystem
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG
enabled
Reported by: yaron nahum
* [028033e5a8] Richard Mudgett -- res/ari/config.c: Fix conf_alloc()
object init.
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
- add ECDH support and fallback to prime256v1
Reported by: Stefan EngstrAP:m
* [9a12804e59] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
when enabling PFS.
* [aed068844c] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
Category: Core/General
ASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes
erroneous duplicate RTCP messages; other potential scheduling issues in
chan_sip/chan_skinny
Reported by: Matt Jordan
* [10e790f81a] Matt Jordan -- res/res_rtp_asterisk: Fix assignment after
ao2 decrement
* [754daeca0a] Matt Jordan -- Fix improper usage of scheduler exposed by
5c713fdf18f
ASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG
enabled
Reported by: yaron nahum
* [028033e5a8] Richard Mudgett -- res/ari/config.c: Fix conf_alloc()
object init.
ASTERISK-25418: On-hold channels redirected out of a bridge appear to
still be on hold
Reported by: Mark Michelson
* [629458d349] Mark Michelson -- Do not swallow frames on channels
leaving bridges.
ASTERISK-25355: sched: ast_sched_del may return prematurely due to
spurious wakeup
Reported by: Joshua Colp
* [85e1cb51b2] Joshua Colp -- sched: ast_sched_del may return
prematurely due to spurious wakeup
Category: Core/Logging
ASTERISK-25305: Dynamic logger channels can be added multiple times
Reported by: Mark Michelson
* [f050fa76eb] Mark Michelson -- logger: Prevent duplicate dynamic
channels from being added.
ASTERISK-25407: Asterisk fails to log to multiple syslog destinations
Reported by: Elazar Broad
* [ec514ad64d] Elazar Broad -- core/logging: Fix logging to more than
one syslog channel
Category: Core/PBX
ASTERISK-25394: pbx: Incorrect device and presence state when changing
hint details
Reported by: Joshua Colp
* [2bd27d1222] Joshua Colp -- pbx: Update device and presence state when
changing a hint extension.
ASTERISK-25367: pbx: Long pattern match hints may cause "core show hints"
to crash
Reported by: Joshua Colp
* [cc1363209e] Joshua Colp -- pbx: Fix crash when issuing "core show
hints" with long pattern match.
ASTERISK-25362: Deadlock due to presence state callback
Reported by: Mark Michelson
* [03fe79f29e] Mark Michelson -- Fix deadlock on presence state changes.
Category: Core/RTP
ASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett
* [aeeb170fc4] Richard Mudgett -- rtp_engine.c: Fix performance issue
with several channel drivers that use RTP.
* [84262749d2] Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal
crash potential.
Category: Resources/res_ari
ASTERISK-25325: ARI PUT reload chan_sip HTTP response 404
Reported by: Rodrigo Ramirez Norambuena
* [865377fc38] Rodrigo RamArez Norambuena -- chan_sip.c: Validation on
module reload
Category: Resources/res_http_websocket
ASTERISK-25312: res_http_websocket: Terminate connection on fatal cases
Reported by: Joshua Colp
* [b4e9416138] Joshua Colp -- res_http_websocket: Forcefully terminate
on write errors.
Category: Resources/res_parking
ASTERISK-25369: res_parking: ParkAndAnnounce - Inheritable variables
aren't applied to the announcer channel
Reported by: Jonathan Rose
* [fbf720db91] Jonathan Rose -- ParkAndAnnounce: Add variable
inheritance
Category: Resources/res_pjsip
ASTERISK-25295: res_pjsip crash - pjsip_uri_get_uri at
/usr/include/pjsip/sip_uri.h
Reported by: Dmitriy Serov
* [5469caa9dd] Joshua Colp -- res_pjsip: Use hash for contact object
identity instead of Contact URI.
* [a676ba2aad] Joshua Colp -- taskprocessor: Fix race condition between
unreferencing and finding.
ASTERISK-25381: res_pjsip: AoRs deleted via ARI (or other mechanism) do
not destroy their related contacts
Reported by: Matt Jordan
* [c3e6debdb9] Matt Jordan -- res/res_pjsip: Purge contacts when an AoR
is deleted
ASTERISK-25339: res_pjsip: Empty "auth" sections from non-config
backgrounds are interpreted as valid
Reported by: Matt Jordan
* [bc6fe07f5c] Matt Jordan -- res_pjsip/pjsip_configuration: Disregard
empty auth values
ASTERISK-25304: res_pjsip: XML sanitization may write past buffer
Reported by: Joshua Colp
* [8521a86367] Joshua Colp -- res_pjsip: Ensure sanitized XML is NULL
terminated.
Category: Resources/res_pjsip_nat
ASTERISK-25387: res_pjsip_nat: Malformed REGISTER request causes NAT'd
Contact header to not be rewritten
Reported by: Matt Jordan
* [1dd0e220bf] Matt Jordan -- res/res_pjsip_nat: Ignore REGISTER
requests when looking for a Record-Route
Category: Resources/res_pjsip_pubsub
ASTERISK-25306: Persistent subscriptions can save multiple SIP messages at
once, leading to potential crashes.
Reported by: Mark Michelson
* [c126afe18f] Richard Mudgett -- res_pjsip.c: Fix crash from corrupt
saved SUBSCRIBE message.
* [e25569ef95] Mark Michelson -- res_pjsip_pubsub: More accurately
persist packet.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-25356: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may
exist
Reported by: Joshua Colp
* [1b1561f4c8] Joshua Colp -- res_pjsip_sdp_rtp: Fix multiple keepalive
scheduled items.
Category: Resources/res_pjsip_session
ASTERISK-25297: Crashes running
channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
Reported by: Richard Mudgett
* [13eb491e35] Richard Mudgett -- res_pjsip_session.c: Fix crashes seen
when call cancelled.
Category: Resources/res_rtp_asterisk
ASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not
enabled
Reported by: Joshua Colp
* [9913d47697] Joshua Colp -- res_rtp_asterisk: Move "Set role" warning
to be debug.
ASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39
- add ECDH support and fallback to prime256v1
Reported by: Stefan EngstrAP:m
* [9a12804e59] Joshua Colp -- res_rtp_asterisk: Don't leak temporary key
when enabling PFS.
* [aed068844c] Mark Duncan -- res/res_rtp_asterisk: Add ECDH support
Category: Tests/testsuite
ASTERISK-25318:
tests/rest_api/applications/subscribe-endpoint/nominal/resource:
Sporadically failing
Reported by: Joshua Colp
* [c2c7319082] Joshua Colp -- res_pjsip_session: Don't invoke session
supplements twice for BYE requests.
ASTERISK-25292: Testuite:
tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
Reported by: Kevin Harwell
* [10ba72a927] Mark Michelson -- Add a test event for inband ringing.
Improvement
Category: Core/General
ASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to
PTHREAD_EXPLICIT_SCHED
Reported by: Guido Falsi
* [4ed9c9a280] Guido Falsi -- Core/General: Add #ifdef needed on
FreeBSD.
Category: Resources/res_ari_applications
ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan
* [90165e306d] Matt Jordan -- res/res_stasis: Fix accidental
subscription to 'all' bridge topic
* [b50e372394] Matt Jordan -- ARI: Add events for Contact and Peer
Status changes
* [3502c0431d] Matt Jordan -- res/res_stasis_device_state: Allow for
subscribing to 'all' device state
* [4c9f613309] Matt Jordan -- ARI: Add the ability to subscribe to all
events
Category: Resources/res_ari_bridges
ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan
* [90165e306d] Matt Jordan -- res/res_stasis: Fix accidental
subscription to 'all' bridge topic
* [b50e372394] Matt Jordan -- ARI: Add events for Contact and Peer
Status changes
* [3502c0431d] Matt Jordan -- res/res_stasis_device_state: Allow for
subscribing to 'all' device state
* [4c9f613309] Matt Jordan -- ARI: Add the ability to subscribe to all
events
----------------------------------------------------------------------
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: Addons/chan_ooh323
ASTERISK-25299: RTP port leaks with incoming OOH323 calls
Reported by: Alexandr Dranchuk
* [480c443e26] Alexander Anikin -- chan_ooh323: call
ast_rtp_instance_stop on ooh323_destroy
Category: General
ASTERISK-25323: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG
Reported by: Scott Griepentrog
* [c94f46080f] Scott Griepentrog -- CHAOS: avoid crash if string create
fails
* [4cc59533b9] Richard Mudgett -- CHAOS: res_pjsip_diversion avoid crash
if allocation fails
* [fb6b5c684b] Scott Griepentrog -- PJSIP: avoid crash when getting rtp
peer
* [f72f9ceefc] Scott Griepentrog -- pjsip: avoid possible crash req_caps
allocation failure
* [6862c2a167] Scott Griepentrog -- Chaos: handle failed allocation in
get_media_encryption_type
* [f1cd636658] Scott Griepentrog -- Chaos: make hangup NULL tolerant
* [ab373f2cef] Scott Griepentrog -- CHAOS: prevent sorcery object with
null id
Category: Resources/res_hep_rtcp
ASTERISK-25352: res_hep_rtcp correlation_id is different then res_hep
Reported by: Kevin Scott Adams
* [78d0b9d97e] Matt Jordan -- channels/pjsip/dialplan_functions: Add an
option for extracting the SIP call-id
Category: pjproject/pjsip
ASTERISK-24602: Unable to call WebRTC client via wss on chan_pjsip
Reported by: Oleg Kozlov
* [d32e516c7c] Martin Tomec -- res/pjsip: Mark WSS transport as secure
Improvement
----------------------------------------------------------------------
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 |
|------------+-----------------+-----------------------------------------|
| cc0eff5651 | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| 8cd191b885 | Kevin Harwell | .version: Update for 13.6.0 |
|------------+-----------------+-----------------------------------------|
| a3777c24fd | Kevin Harwell | .lastclean: Update for 13.6.0 |
|------------+-----------------+-----------------------------------------|
| 68121cef21 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0 |
|------------+-----------------+-----------------------------------------|
| d72dab4f40 | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 9da83dbd15 | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 8c60f9326c | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| 316d47755b | Kevin Harwell | .version: Update for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 74a86d0a72 | Kevin Harwell | .lastclean: Update for 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| 4c39bea6f0 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc3 |
|------------+-----------------+-----------------------------------------|
| c3521e9469 | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| a44f6aa046 | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| dd74af7e46 | Kevin Harwell | Release summaries: Remove previous |
| | | versions |
|------------+-----------------+-----------------------------------------|
| a11a78ca34 | Kevin Harwell | .version: Update for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| 570329ec8a | Kevin Harwell | .lastclean: Update for 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| 51c9ff47f6 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc2 |
|------------+-----------------+-----------------------------------------|
| a0fb436eda | Kevin Harwell | ChangeLog: Updated for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| bba1c4066b | Kevin Harwell | Release summaries: Add summaries for |
| | | 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| 82c4aecdbb | Kevin Harwell | .version: Update for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| bc18db7388 | Kevin Harwell | .lastclean: Update for 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| b9c53f95e3 | Kevin Harwell | realtime: Add database scripts for |
| | | 13.6.0-rc1 |
|------------+-----------------+-----------------------------------------|
| d30939b6e8 | Kevin Harwell | ARI: Changed version from 1.8.0 to |
| | | 1.9.0 |
|------------+-----------------+-----------------------------------------|
| 5f19c9bade | Richard Mudgett | res/ari/config.c: Fix user sort compare |
| | | function. |
|------------+-----------------+-----------------------------------------|
| 3a85764039 | Richard Mudgett | res/ari/config.c: Optimize conf_alloc() |
| | | object init. |
|------------+-----------------+-----------------------------------------|
| bbeda190c3 | Richard Mudgett | app_dial.c: Remove some no-op code. |
|------------+-----------------+-----------------------------------------|
| fe5077b1f8 | Mark Michelson | res_pjsip_pubsub: Eliminate race during |
| | | initial NOTIFY. |
|------------+-----------------+-----------------------------------------|
| 5c713fdf18 | Mark Michelson | scheduler: Use queue for allocating |
| | | sched IDs. |
|------------+-----------------+-----------------------------------------|
| | | res_pjsip_pubsub.c: Mark |
| e75aff53e6 | Richard Mudgett | ast_sip_create_subscription() as not |
| | | used. |
|------------+-----------------+-----------------------------------------|
| 4d91d01df1 | Richard Mudgett | res_pjsip_pubsub.c: Add some |
| | | notification comments. |
|------------+-----------------+-----------------------------------------|
| f36a9d1221 | Richard Mudgett | res_pjsip_pubsub.c: Set dlg_status code |
| | | instead of sending SIP response. |
|------------+-----------------+-----------------------------------------|
| 94582f8fab | Richard Mudgett | res_pjsip_pubsub.c: Fix off-nominal |
| | | memory leak. |
|------------+-----------------+-----------------------------------------|
| 8b3ed52239 | Richard Mudgett | res_pjsip_pubsub.c: Fix one byte buffer |
| | | overrun error. |
|------------+-----------------+-----------------------------------------|
| 4329bd1e4c | Richard Mudgett | res_pjsip_pubsub.c: Use ast_alloca() |
| | | instead of alloca(). |
|------------+-----------------+-----------------------------------------|
| a456a20ecf | Richard Mudgett | res_pjsip_pubsub.c: Add missing error |
| | | return in load_module(). |
|------------+-----------------+-----------------------------------------|
| f58f4c6e27 | Richard Mudgett | res_pjsip/location.c: Use the builtin |
| | | ao2_callback() match function instead. |
|------------+-----------------+-----------------------------------------|
| | | main/config_options: Check for |
| 4eedd9ef9d | Matt Jordan | existance of internal object before |
| | | derefing |
|------------+-----------------+-----------------------------------------|
| 695f26cbb7 | David M. Lee | res_rtp_asterisk: Add more ICE |
| | | debugging |
|------------+-----------------+-----------------------------------------|
| 61c6c6aa6c | David M. Lee | Fix when remote candidates exceed |
| | | PJ_ICE_MAX_CAND |
|------------+-----------------+-----------------------------------------|
| ad9cb6c2ce | Mark Michelson | res_pjsip: Fix contact refleak on |
| | | stateful responses. |
|------------+-----------------+-----------------------------------------|
| 7c4d0c3506 | Joshua Colp | res_pjsip_pubsub: On recreated notify |
| | | fail deleted sub_tree is referenced |
|------------+-----------------+-----------------------------------------|
| 0582776f7f | Richard Mudgett | ari/ari_websockets.c: Fix ast_debug |
| | | parameter type mismatch. |
|------------+-----------------+-----------------------------------------|
| 77518d5434 | Richard Mudgett | res_http_websocket.c: Fix some off |
| | | nominal path cleanup. |
|------------+-----------------+-----------------------------------------|
| c61547fee6 | Richard Mudgett | res_ari.c: Add missing off nominal |
| | | unlock and remove a RAII_VAR(). |
|------------+-----------------+-----------------------------------------|
| bd867cd078 | Richard Mudgett | app_queue.c: Extract some functions for |
| | | simpler code. |
|------------+-----------------+-----------------------------------------|
| ded51e3d77 | Richard Mudgett | app_queue.c: Fix error checking in |
| | | QUEUE_MEMBER() read. |
|------------+-----------------+-----------------------------------------|
| b719f56c72 | Mark Michelson | res_pjsip_sdp_rtp: Restore removed NULL |
| | | check. |
|------------+-----------------+-----------------------------------------|
| cea5dc7b8a | Richard Mudgett | audiohook.c: Simplify variable usage in |
| | | audiohook_read_frame_both(). |
|------------+-----------------+-----------------------------------------|
| e18c300550 | Joshua Colp | res_http_websocket: When shutting down |
| | | a session don't close closed socket |
|------------+-----------------+-----------------------------------------|
| 8e194047ac | Matt Jordan | res/res_format_attr_silk: Expose format |
| | | attributes to other modules |
|------------+-----------------+-----------------------------------------|
| a0f451c35e | Matt Jordan | main/format: Add an API call for |
| | | retrieving format attributes |
|------------+-----------------+-----------------------------------------|
| 26f0559a94 | David M. Lee | Replace htobe64 with htonll |
|------------+-----------------+-----------------------------------------|
| 27dc2094e9 | Mark Michelson | res_http_websocket: Debug write |
| | | lengths. |
|------------+-----------------+-----------------------------------------|
| 39cc28f6ea | Mark Michelson | res_http_websocket: Avoid passing |
| | | strlen() to ast_websocket_write(). |
|------------+-----------------+-----------------------------------------|
| 1519eb44a7 | Richard Mudgett | rtp_engine.c: Must protect |
| | | mime_types_len with mime_types_lock. |
|------------+-----------------+-----------------------------------------|
| a93b7a927c | Richard Mudgett | res_pjsip_sdp_rtp.c: Fix processing |
| | | wrong SDP media list. |
|------------+-----------------+-----------------------------------------|
| 741fa0d26d | Richard Mudgett | res_pjsip_sdp_rtp.c: Fixup some |
| | | whitespace. |
|------------+-----------------+-----------------------------------------|
| 89b21fd9a3 | Richard Mudgett | rtp_engine.h: No sense allowing payload |
| | | types larger than RFC allows. |
|------------+-----------------+-----------------------------------------|
| 7427c7f13b | Richard Mudgett | rtp_engine.c: Minor tweaks. |
|------------+-----------------+-----------------------------------------|
| e20f435b60 | Richard Mudgett | rtp_engine.h: Misc comment fixes. |
|------------+-----------------+-----------------------------------------|
| bc5d7f9c37 | Richard Mudgett | chan_sip.c: Tweak glue->update_peer() |
| | | parameter nil value. |
|------------+-----------------+-----------------------------------------|
| 48698a5e21 | Mark Michelson | res_http_websocket: Properly encode 64 |
| | | bit payload |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
asterisk-13.5.0-summary.html | 441 --
asterisk-13.5.0-summary.txt | 1141 ------
b/.version | 2
b/CHANGES | 35
b/ChangeLog | 1884 +++++++++-
b/Makefile | 1
b/UPGRADE.txt | 6
b/addons/chan_ooh323.c | 1
b/addons/ooh323c/src/ooq931.c | 6
b/apps/app_dial.c | 156
b/apps/app_page.c | 28
b/apps/app_queue.c | 412 +-
b/apps/app_record.c | 3
b/bridges/bridge_holding.c | 6
b/channels/chan_dahdi.c | 55
b/channels/chan_pjsip.c | 18
b/channels/chan_sip.c | 78
b/channels/chan_skinny.c | 26
b/channels/pjsip/dialplan_functions.c | 5
b/channels/sip/include/security_events.h | 3
b/channels/sip/security_events.c | 5
b/codecs/codec_gsm.c | 29
b/codecs/codec_ilbc.c | 28
b/codecs/codec_lpc10.c | 41
b/codecs/codec_speex.c | 60
b/configure | 63
b/configure.ac | 6
b/contrib/ast-db-manage/config/versions/154177371065_add_default_from_user.py | 22
b/contrib/realtime/mssql/mssql_config.sql | 8
b/contrib/realtime/mysql/mysql_config.sql | 6
b/contrib/realtime/oracle/oracle_config.sql | 8
b/contrib/realtime/postgresql/postgresql_config.sql | 6
b/contrib/scripts/astversion | 536 ++
b/contrib/scripts/install_prereq | 2
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/format.h | 23
b/include/asterisk/logger.h | 44
b/include/asterisk/res_pjsip.h | 40
b/include/asterisk/res_pjsip_session.h | 6
b/include/asterisk/rtp_engine.h | 18
b/include/asterisk/stasis_app.h | 15
b/main/audiohook.c | 43
b/main/bridge_channel.c | 7
b/main/channel.c | 9
b/main/config_options.c | 4
b/main/dial.c | 25
b/main/endpoints.c | 3
b/main/format.c | 11
b/main/logger.c | 272 +
b/main/pbx.c | 146
b/main/rtp_engine.c | 153
b/main/sched.c | 143
b/main/sorcery.c | 4
b/main/stasis_endpoints.c | 78
b/main/taskprocessor.c | 12
b/main/translate.c | 55
b/main/utils.c | 4
b/res/ari/ari_model_validators.c | 445 ++
b/res/ari/ari_model_validators.h | 118
b/res/ari/ari_websockets.c | 9
b/res/ari/config.c | 72
b/res/ari/resource_asterisk.c | 127
b/res/ari/resource_asterisk.h | 63
b/res/ari/resource_events.c | 24
b/res/ari/resource_events.h | 2
b/res/parking/parking_applications.c | 65
b/res/res_ari.c | 19
b/res/res_ari_asterisk.c | 300 +
b/res/res_ari_events.c | 6
b/res/res_config_sqlite.c | 8
b/res/res_format_attr_silk.c | 24
b/res/res_http_websocket.c | 80
b/res/res_pjsip.c | 131
b/res/res_pjsip/config_global.c | 18
b/res/res_pjsip/location.c | 51
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/presence_xml.c | 31
b/res/res_pjsip_diversion.c | 4
b/res/res_pjsip_multihomed.c | 23
b/res/res_pjsip_nat.c | 2
b/res/res_pjsip_pidf_digium_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 80
b/res/res_pjsip_sdp_rtp.c | 70
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_t38.c | 15
b/res/res_pjsip_transport_websocket.c | 5
b/res/res_rtp_asterisk.c | 79
b/res/res_stasis.c | 45
b/res/res_stasis_device_state.c | 54
b/res/stasis/app.c | 335 +
b/res/stasis/app.h | 15
b/res/stasis/messaging.c | 44
b/rest-api-templates/ari_model_validators.c.mustache | 2
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 130
b/rest-api/api-docs/bridges.json | 2
b/rest-api/api-docs/channels.json | 2
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 109
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/tests/test_core_format.c | 57
106 files changed, 6435 insertions(+), 2535 deletions(-)

View File

@@ -4318,7 +4318,7 @@ void sip_scheddestroy(struct sip_pvt *p, int ms)
}
p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, dialog_ref(p, "setting ref as passing into ast_sched_add for __sip_autodestruct"));
if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > 0) {
if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > -1) {
stop_session_timer(p);
}
}
@@ -22739,7 +22739,7 @@ static void check_pendings(struct sip_pvt *p)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
/* if we can't REINVITE, hold it for later */
if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) {
if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > -1) {
ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
} else {
ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);

View File

@@ -4863,7 +4863,7 @@ static int skinny_dialer_cb(const void *data)
{
struct skinny_subchannel *sub = (struct skinny_subchannel *)data;
SKINNY_DEBUG(DEBUG_SUB, 3, "Sub %u - Dialer called from SCHED %d\n", sub->callid, sub->dialer_sched);
sub->dialer_sched = 0;
sub->dialer_sched = -1;
skinny_dialer(sub, 1);
return 0;
}
@@ -4872,7 +4872,7 @@ static int skinny_autoanswer_cb(const void *data)
{
struct skinny_subchannel *sub = (struct skinny_subchannel *)data;
skinny_locksub(sub);
sub->aa_sched = 0;
sub->aa_sched = -1;
setsubstate(sub, SKINNY_CONNECTED);
skinny_unlocksub(sub);
return 0;
@@ -4882,7 +4882,7 @@ static int skinny_cfwd_cb(const void *data)
{
struct skinny_subchannel *sub = (struct skinny_subchannel *)data;
struct skinny_line *l = sub->line;
sub->cfwd_sched = 0;
sub->cfwd_sched = -1;
SKINNY_DEBUG(DEBUG_SUB, 3, "Sub %u - CFWDNOANS to %s.\n", sub->callid, l->call_forward_noanswer);
ast_channel_call_forward_set(sub->owner, l->call_forward_noanswer);
ast_queue_control(sub->owner, AST_CONTROL_REDIRECTING);
@@ -4924,7 +4924,7 @@ static int skinny_call(struct ast_channel *ast, const char *dest, int timeout)
skinny_locksub(sub);
AST_LIST_TRAVERSE(ast_channel_varshead(ast), current, entries) {
if (!(strcmp(ast_var_name(current), "SKINNY_AUTOANSWER"))) {
if (d->hookstate == SKINNY_ONHOOK && !sub->aa_sched) {
if (d->hookstate == SKINNY_ONHOOK && sub->aa_sched < 0) {
char buf[24];
int aatime;
char *stringp = buf, *curstr;
@@ -5389,9 +5389,9 @@ static struct ast_channel *skinny_new(struct skinny_line *l, struct skinny_subli
sub->xferor = 0;
sub->related = NULL;
sub->calldirection = direction;
sub->aa_sched = 0;
sub->dialer_sched = 0;
sub->cfwd_sched = 0;
sub->aa_sched = -1;
sub->dialer_sched = -1;
sub->cfwd_sched = -1;
sub->dialType = DIALTYPE_NORMAL;
sub->getforward = 0;
@@ -5550,17 +5550,17 @@ static void setsubstate(struct skinny_subchannel *sub, int state)
if (sub->dialer_sched) {
skinny_sched_del(sub->dialer_sched, sub);
sub->dialer_sched = 0;
sub->dialer_sched = -1;
}
if (state != SUBSTATE_RINGIN && sub->aa_sched) {
skinny_sched_del(sub->aa_sched, sub);
sub->aa_sched = 0;
sub->aa_sched = -1;
sub->aa_beep = 0;
sub->aa_mute = 0;
}
if (sub->cfwd_sched) {
if (sub->cfwd_sched > -1) {
if (state == SUBSTATE_CONNECTED) {
if (skinny_sched_del(sub->cfwd_sched, sub)) {
SKINNY_DEBUG(DEBUG_SUB, 3, "Sub %u - trying to change state from %s to %s, but already forwarded because no answer.\n",
@@ -5568,7 +5568,7 @@ static void setsubstate(struct skinny_subchannel *sub, int state)
skinny_unlocksub(sub);
return;
}
sub->cfwd_sched = 0;
sub->cfwd_sched = -1;
} else if (state == SUBSTATE_ONHOOK) {
skinny_sched_del(sub->cfwd_sched, sub);
}
@@ -6240,7 +6240,7 @@ static int handle_keypad_button_message(struct skinny_req *req, struct skinnyses
if ((sub->owner && ast_channel_state(sub->owner) < AST_STATE_UP)) {
if (sub->dialer_sched && !skinny_sched_del(sub->dialer_sched, sub)) {
SKINNY_DEBUG(DEBUG_SUB, 3, "Sub %u - Got a digit and not timed out, so try dialing\n", sub->callid);
sub->dialer_sched = 0;
sub->dialer_sched = -1;
len = strlen(sub->exten);
if (len == 0) {
transmit_stop_tone(d, l->instance, sub->callid);
@@ -7077,7 +7077,7 @@ static int handle_soft_key_event_message(struct skinny_req *req, struct skinnyse
d->name, instance, callreference);
if (sub->dialer_sched && !skinny_sched_del(sub->dialer_sched, sub)) {
size_t len;
sub->dialer_sched = 0;
sub->dialer_sched = -1;
len = strlen(sub->exten);
if (len > 0) {
sub->exten[len-1] = '\0';

View File

@@ -0,0 +1,42 @@
BEGIN TRANSACTION;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
GO
-- Running upgrade None -> 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;

File diff suppressed because it is too large Load Diff

View File

@@ -0,0 +1,48 @@
BEGIN TRANSACTION;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
GO
-- Running upgrade None -> 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
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE;
GO
INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5');
GO
COMMIT;

View File

@@ -0,0 +1,32 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 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');

View File

@@ -0,0 +1,769 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 4da0c5f79a9c
CREATE TABLE sippeers (
id INTEGER NOT NULL AUTO_INCREMENT,
name VARCHAR(40) NOT NULL,
ipaddr VARCHAR(45),
port INTEGER,
regseconds INTEGER,
defaultuser VARCHAR(40),
fullcontact VARCHAR(80),
regserver VARCHAR(20),
useragent VARCHAR(20),
lastms INTEGER,
host VARCHAR(40),
type ENUM('friend','user','peer'),
context VARCHAR(40),
permit VARCHAR(95),
deny VARCHAR(95),
secret VARCHAR(40),
md5secret VARCHAR(40),
remotesecret VARCHAR(40),
transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp'),
dtmfmode ENUM('rfc2833','info','shortinfo','inband','auto'),
directmedia ENUM('yes','no','nonat','update'),
nat VARCHAR(29),
callgroup VARCHAR(40),
pickupgroup VARCHAR(40),
language VARCHAR(40),
disallow VARCHAR(200),
allow VARCHAR(200),
insecure VARCHAR(40),
trustrpid ENUM('yes','no'),
progressinband ENUM('yes','no','never'),
promiscredir ENUM('yes','no'),
useclientcode ENUM('yes','no'),
accountcode VARCHAR(40),
setvar VARCHAR(200),
callerid VARCHAR(40),
amaflags VARCHAR(40),
callcounter ENUM('yes','no'),
busylevel INTEGER,
allowoverlap ENUM('yes','no'),
allowsubscribe ENUM('yes','no'),
videosupport ENUM('yes','no'),
maxcallbitrate INTEGER,
rfc2833compensate ENUM('yes','no'),
mailbox VARCHAR(40),
`session-timers` ENUM('accept','refuse','originate'),
`session-expires` INTEGER,
`session-minse` INTEGER,
`session-refresher` ENUM('uac','uas'),
t38pt_usertpsource VARCHAR(40),
regexten VARCHAR(40),
fromdomain VARCHAR(40),
fromuser VARCHAR(40),
qualify VARCHAR(40),
defaultip VARCHAR(45),
rtptimeout INTEGER,
rtpholdtimeout INTEGER,
sendrpid ENUM('yes','no'),
outboundproxy VARCHAR(40),
callbackextension VARCHAR(40),
timert1 INTEGER,
timerb INTEGER,
qualifyfreq INTEGER,
constantssrc ENUM('yes','no'),
contactpermit VARCHAR(95),
contactdeny VARCHAR(95),
usereqphone ENUM('yes','no'),
textsupport ENUM('yes','no'),
faxdetect ENUM('yes','no'),
buggymwi ENUM('yes','no'),
auth VARCHAR(40),
fullname VARCHAR(40),
trunkname VARCHAR(40),
cid_number VARCHAR(40),
callingpres ENUM('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib'),
mohinterpret VARCHAR(40),
mohsuggest VARCHAR(40),
parkinglot VARCHAR(40),
hasvoicemail ENUM('yes','no'),
subscribemwi ENUM('yes','no'),
vmexten VARCHAR(40),
autoframing ENUM('yes','no'),
rtpkeepalive INTEGER,
`call-limit` INTEGER,
g726nonstandard ENUM('yes','no'),
ignoresdpversion ENUM('yes','no'),
allowtransfer ENUM('yes','no'),
dynamic ENUM('yes','no'),
path VARCHAR(256),
supportpath ENUM('yes','no'),
PRIMARY KEY (id),
UNIQUE (name)
);
CREATE INDEX sippeers_name ON sippeers (name);
CREATE INDEX sippeers_name_host ON sippeers (name, host);
CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port);
CREATE INDEX sippeers_host_port ON sippeers (host, port);
CREATE TABLE iaxfriends (
id INTEGER NOT NULL AUTO_INCREMENT,
name VARCHAR(40) NOT NULL,
type ENUM('friend','user','peer'),
username VARCHAR(40),
mailbox VARCHAR(40),
secret VARCHAR(40),
dbsecret VARCHAR(40),
context VARCHAR(40),
regcontext VARCHAR(40),
host VARCHAR(40),
ipaddr VARCHAR(40),
port INTEGER,
defaultip VARCHAR(20),
sourceaddress VARCHAR(20),
mask VARCHAR(20),
regexten VARCHAR(40),
regseconds INTEGER,
accountcode VARCHAR(20),
mohinterpret VARCHAR(20),
mohsuggest VARCHAR(20),
inkeys VARCHAR(40),
outkeys VARCHAR(40),
language VARCHAR(10),
callerid VARCHAR(100),
cid_number VARCHAR(40),
sendani ENUM('yes','no'),
fullname VARCHAR(40),
trunk ENUM('yes','no'),
auth VARCHAR(20),
maxauthreq INTEGER,
requirecalltoken ENUM('yes','no','auto'),
encryption ENUM('yes','no','aes128'),
transfer ENUM('yes','no','mediaonly'),
jitterbuffer ENUM('yes','no'),
forcejitterbuffer ENUM('yes','no'),
disallow VARCHAR(200),
allow VARCHAR(200),
codecpriority VARCHAR(40),
qualify VARCHAR(10),
qualifysmoothing ENUM('yes','no'),
qualifyfreqok VARCHAR(10),
qualifyfreqnotok VARCHAR(10),
timezone VARCHAR(20),
adsi ENUM('yes','no'),
amaflags VARCHAR(20),
setvar VARCHAR(200),
PRIMARY KEY (id),
UNIQUE (name)
);
CREATE INDEX iaxfriends_name ON iaxfriends (name);
CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host);
CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port);
CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port);
CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port);
CREATE TABLE voicemail (
uniqueid INTEGER NOT NULL AUTO_INCREMENT,
context VARCHAR(80) NOT NULL,
mailbox VARCHAR(80) NOT NULL,
password VARCHAR(80) NOT NULL,
fullname VARCHAR(80),
alias VARCHAR(80),
email VARCHAR(80),
pager VARCHAR(80),
attach ENUM('yes','no'),
attachfmt VARCHAR(10),
serveremail VARCHAR(80),
language VARCHAR(20),
tz VARCHAR(30),
deletevoicemail ENUM('yes','no'),
saycid ENUM('yes','no'),
sendvoicemail ENUM('yes','no'),
review ENUM('yes','no'),
tempgreetwarn ENUM('yes','no'),
operator ENUM('yes','no'),
envelope ENUM('yes','no'),
sayduration INTEGER,
forcename ENUM('yes','no'),
forcegreetings ENUM('yes','no'),
callback VARCHAR(80),
dialout VARCHAR(80),
exitcontext VARCHAR(80),
maxmsg INTEGER,
volgain NUMERIC(5, 2),
imapuser VARCHAR(80),
imappassword VARCHAR(80),
imapserver VARCHAR(80),
imapport VARCHAR(8),
imapflags VARCHAR(80),
stamp DATETIME,
PRIMARY KEY (uniqueid)
);
CREATE INDEX voicemail_mailbox ON voicemail (mailbox);
CREATE INDEX voicemail_context ON voicemail (context);
CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context);
CREATE INDEX voicemail_imapuser ON voicemail (imapuser);
CREATE TABLE meetme (
bookid INTEGER NOT NULL AUTO_INCREMENT,
confno VARCHAR(80) NOT NULL,
starttime DATETIME,
endtime DATETIME,
pin VARCHAR(20),
adminpin VARCHAR(20),
opts VARCHAR(20),
adminopts VARCHAR(20),
recordingfilename VARCHAR(80),
recordingformat VARCHAR(10),
maxusers INTEGER,
members INTEGER NOT NULL,
PRIMARY KEY (bookid)
);
CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime);
CREATE TABLE musiconhold (
name VARCHAR(80) NOT NULL,
mode ENUM('custom','files','mp3nb','quietmp3nb','quietmp3'),
directory VARCHAR(255),
application VARCHAR(255),
digit VARCHAR(1),
sort VARCHAR(10),
format VARCHAR(10),
stamp DATETIME,
PRIMARY KEY (name)
);
INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c');
-- Running upgrade 4da0c5f79a9c -> 43956d550a44
CREATE TABLE ps_endpoints (
id VARCHAR(40) NOT NULL,
transport VARCHAR(40),
aors VARCHAR(200),
auth VARCHAR(40),
context VARCHAR(40),
disallow VARCHAR(200),
allow VARCHAR(200),
direct_media ENUM('yes','no'),
connected_line_method ENUM('invite','reinvite','update'),
direct_media_method ENUM('invite','reinvite','update'),
direct_media_glare_mitigation ENUM('none','outgoing','incoming'),
disable_direct_media_on_nat ENUM('yes','no'),
dtmf_mode ENUM('rfc4733','inband','info'),
external_media_address VARCHAR(40),
force_rport ENUM('yes','no'),
ice_support ENUM('yes','no'),
identify_by ENUM('username'),
mailboxes VARCHAR(40),
moh_suggest VARCHAR(40),
outbound_auth VARCHAR(40),
outbound_proxy VARCHAR(40),
rewrite_contact ENUM('yes','no'),
rtp_ipv6 ENUM('yes','no'),
rtp_symmetric ENUM('yes','no'),
send_diversion ENUM('yes','no'),
send_pai ENUM('yes','no'),
send_rpid ENUM('yes','no'),
timers_min_se INTEGER,
timers ENUM('forced','no','required','yes'),
timers_sess_expires INTEGER,
callerid VARCHAR(40),
callerid_privacy ENUM('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable'),
callerid_tag VARCHAR(40),
`100rel` ENUM('no','required','yes'),
aggregate_mwi ENUM('yes','no'),
trust_id_inbound ENUM('yes','no'),
trust_id_outbound ENUM('yes','no'),
use_ptime ENUM('yes','no'),
use_avpf ENUM('yes','no'),
media_encryption ENUM('no','sdes','dtls'),
inband_progress ENUM('yes','no'),
call_group VARCHAR(40),
pickup_group VARCHAR(40),
named_call_group VARCHAR(40),
named_pickup_group VARCHAR(40),
device_state_busy_at INTEGER,
fax_detect ENUM('yes','no'),
t38_udptl ENUM('yes','no'),
t38_udptl_ec ENUM('none','fec','redundancy'),
t38_udptl_maxdatagram INTEGER,
t38_udptl_nat ENUM('yes','no'),
t38_udptl_ipv6 ENUM('yes','no'),
tone_zone VARCHAR(40),
language VARCHAR(40),
one_touch_recording ENUM('yes','no'),
record_on_feature VARCHAR(40),
record_off_feature VARCHAR(40),
rtp_engine VARCHAR(40),
allow_transfer ENUM('yes','no'),
allow_subscribe ENUM('yes','no'),
sdp_owner VARCHAR(40),
sdp_session VARCHAR(40),
tos_audio INTEGER,
tos_video INTEGER,
cos_audio INTEGER,
cos_video INTEGER,
sub_min_expiry INTEGER,
from_domain VARCHAR(40),
from_user VARCHAR(40),
mwi_fromuser VARCHAR(40),
dtls_verify VARCHAR(40),
dtls_rekey VARCHAR(40),
dtls_cert_file VARCHAR(200),
dtls_private_key VARCHAR(200),
dtls_cipher VARCHAR(200),
dtls_ca_file VARCHAR(200),
dtls_ca_path VARCHAR(200),
dtls_setup ENUM('active','passive','actpass'),
srtp_tag_32 ENUM('yes','no'),
UNIQUE (id)
);
CREATE INDEX ps_endpoints_id ON ps_endpoints (id);
CREATE TABLE ps_auths (
id VARCHAR(40) NOT NULL,
auth_type ENUM('md5','userpass'),
nonce_lifetime INTEGER,
md5_cred VARCHAR(40),
password VARCHAR(80),
realm VARCHAR(40),
username VARCHAR(40),
UNIQUE (id)
);
CREATE INDEX ps_auths_id ON ps_auths (id);
CREATE TABLE ps_aors (
id VARCHAR(40) NOT NULL,
contact VARCHAR(40),
default_expiration INTEGER,
mailboxes VARCHAR(80),
max_contacts INTEGER,
minimum_expiration INTEGER,
remove_existing ENUM('yes','no'),
qualify_frequency INTEGER,
authenticate_qualify ENUM('yes','no'),
UNIQUE (id)
);
CREATE INDEX ps_aors_id ON ps_aors (id);
CREATE TABLE ps_contacts (
id VARCHAR(40) NOT NULL,
uri VARCHAR(40),
expiration_time VARCHAR(40),
qualify_frequency INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_contacts_id ON ps_contacts (id);
CREATE TABLE ps_domain_aliases (
id VARCHAR(40) NOT NULL,
domain VARCHAR(80),
UNIQUE (id)
);
CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id);
CREATE TABLE ps_endpoint_id_ips (
id VARCHAR(40) NOT NULL,
endpoint VARCHAR(40),
`match` VARCHAR(80),
UNIQUE (id)
);
CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id);
UPDATE alembic_version SET version_num='43956d550a44';
-- Running upgrade 43956d550a44 -> 581a4264e537
CREATE TABLE extensions (
id BIGINT NOT NULL AUTO_INCREMENT,
context VARCHAR(40) NOT NULL,
exten VARCHAR(40) NOT NULL,
priority INTEGER NOT NULL,
app VARCHAR(40) NOT NULL,
appdata VARCHAR(256) NOT NULL,
PRIMARY KEY (id, context, exten, priority),
UNIQUE (id)
);
UPDATE alembic_version SET version_num='581a4264e537';
-- Running upgrade 581a4264e537 -> 2fc7930b41b3
CREATE TABLE ps_systems (
id VARCHAR(40) NOT NULL,
timer_t1 INTEGER,
timer_b INTEGER,
compact_headers ENUM('yes','no'),
threadpool_initial_size INTEGER,
threadpool_auto_increment INTEGER,
threadpool_idle_timeout INTEGER,
threadpool_max_size INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_systems_id ON ps_systems (id);
CREATE TABLE ps_globals (
id VARCHAR(40) NOT NULL,
max_forwards INTEGER,
user_agent VARCHAR(40),
default_outbound_endpoint VARCHAR(40),
UNIQUE (id)
);
CREATE INDEX ps_globals_id ON ps_globals (id);
CREATE TABLE ps_transports (
id VARCHAR(40) NOT NULL,
async_operations INTEGER,
bind VARCHAR(40),
ca_list_file VARCHAR(200),
cert_file VARCHAR(200),
cipher VARCHAR(200),
domain VARCHAR(40),
external_media_address VARCHAR(40),
external_signaling_address VARCHAR(40),
external_signaling_port INTEGER,
method ENUM('default','unspecified','tlsv1','sslv2','sslv3','sslv23'),
local_net VARCHAR(40),
password VARCHAR(40),
priv_key_file VARCHAR(200),
protocol ENUM('udp','tcp','tls','ws','wss'),
require_client_cert ENUM('yes','no'),
verify_client ENUM('yes','no'),
verifiy_server ENUM('yes','no'),
tos ENUM('yes','no'),
cos ENUM('yes','no'),
UNIQUE (id)
);
CREATE INDEX ps_transports_id ON ps_transports (id);
CREATE TABLE ps_registrations (
id VARCHAR(40) NOT NULL,
auth_rejection_permanent ENUM('yes','no'),
client_uri VARCHAR(40),
contact_user VARCHAR(40),
expiration INTEGER,
max_retries INTEGER,
outbound_auth VARCHAR(40),
outbound_proxy VARCHAR(40),
retry_interval INTEGER,
forbidden_retry_interval INTEGER,
server_uri VARCHAR(40),
transport VARCHAR(40),
support_path ENUM('yes','no'),
UNIQUE (id)
);
CREATE INDEX ps_registrations_id ON ps_registrations (id);
ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40);
ALTER TABLE ps_endpoints ADD COLUMN redirect_method ENUM('user','uri_core','uri_pjsip');
ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT;
ALTER TABLE ps_endpoints CHANGE mwi_fromuser mwi_from_user VARCHAR(40) NULL;
ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40);
ALTER TABLE ps_contacts ADD COLUMN path TEXT;
ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER;
ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40);
ALTER TABLE ps_aors ADD COLUMN support_path ENUM('yes','no');
UPDATE alembic_version SET version_num='2fc7930b41b3';
-- Running upgrade 2fc7930b41b3 -> 21e526ad3040
ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40);
UPDATE alembic_version SET version_num='21e526ad3040';
-- Running upgrade 21e526ad3040 -> 28887f25a46f
CREATE TABLE queues (
name VARCHAR(128) NOT NULL,
musiconhold VARCHAR(128),
announce VARCHAR(128),
context VARCHAR(128),
timeout INTEGER,
ringinuse ENUM('yes','no'),
setinterfacevar ENUM('yes','no'),
setqueuevar ENUM('yes','no'),
setqueueentryvar ENUM('yes','no'),
monitor_format VARCHAR(8),
membermacro VARCHAR(512),
membergosub VARCHAR(512),
queue_youarenext VARCHAR(128),
queue_thereare VARCHAR(128),
queue_callswaiting VARCHAR(128),
queue_quantity1 VARCHAR(128),
queue_quantity2 VARCHAR(128),
queue_holdtime VARCHAR(128),
queue_minutes VARCHAR(128),
queue_minute VARCHAR(128),
queue_seconds VARCHAR(128),
queue_thankyou VARCHAR(128),
queue_callerannounce VARCHAR(128),
queue_reporthold VARCHAR(128),
announce_frequency INTEGER,
announce_to_first_user ENUM('yes','no'),
min_announce_frequency INTEGER,
announce_round_seconds INTEGER,
announce_holdtime VARCHAR(128),
announce_position VARCHAR(128),
announce_position_limit INTEGER,
periodic_announce VARCHAR(50),
periodic_announce_frequency INTEGER,
relative_periodic_announce ENUM('yes','no'),
random_periodic_announce ENUM('yes','no'),
retry INTEGER,
wrapuptime INTEGER,
penaltymemberslimit INTEGER,
autofill ENUM('yes','no'),
monitor_type VARCHAR(128),
autopause ENUM('yes','no','all'),
autopausedelay INTEGER,
autopausebusy ENUM('yes','no'),
autopauseunavail ENUM('yes','no'),
maxlen INTEGER,
servicelevel INTEGER,
strategy ENUM('ringall','leastrecent','fewestcalls','random','rrmemory','linear','wrandom','rrordered'),
joinempty VARCHAR(128),
leavewhenempty VARCHAR(128),
reportholdtime ENUM('yes','no'),
memberdelay INTEGER,
weight INTEGER,
timeoutrestart ENUM('yes','no'),
defaultrule VARCHAR(128),
timeoutpriority VARCHAR(128),
PRIMARY KEY (name)
);
CREATE TABLE queue_members (
queue_name VARCHAR(80) NOT NULL,
interface VARCHAR(80) NOT NULL,
uniqueid VARCHAR(80) NOT NULL,
membername VARCHAR(80),
state_interface VARCHAR(80),
penalty INTEGER,
paused INTEGER,
PRIMARY KEY (queue_name, interface)
);
UPDATE alembic_version SET version_num='28887f25a46f';
-- Running upgrade 28887f25a46f -> 4c573e7135bd
ALTER TABLE ps_endpoints CHANGE tos_audio tos_audio VARCHAR(10) NULL;
ALTER TABLE ps_endpoints CHANGE tos_video tos_video VARCHAR(10) NULL;
ALTER TABLE ps_transports CHANGE tos tos VARCHAR(10) NULL;
ALTER TABLE ps_endpoints DROP COLUMN cos_audio;
ALTER TABLE ps_endpoints DROP COLUMN cos_video;
ALTER TABLE ps_transports DROP COLUMN cos;
ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER;
ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER;
ALTER TABLE ps_transports ADD COLUMN cos INTEGER;
UPDATE alembic_version SET version_num='4c573e7135bd';
-- Running upgrade 4c573e7135bd -> 3855ee4e5f85
ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40);
ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40);
UPDATE alembic_version SET version_num='3855ee4e5f85';
-- Running upgrade 3855ee4e5f85 -> e96a0b8071c
ALTER TABLE ps_globals CHANGE user_agent user_agent VARCHAR(255) NULL;
ALTER TABLE ps_contacts CHANGE id id VARCHAR(255) NULL;
ALTER TABLE ps_contacts CHANGE uri uri VARCHAR(255) NULL;
ALTER TABLE ps_contacts CHANGE user_agent user_agent VARCHAR(255) NULL;
ALTER TABLE ps_registrations CHANGE client_uri client_uri VARCHAR(255) NULL;
ALTER TABLE ps_registrations CHANGE server_uri server_uri VARCHAR(255) NULL;
UPDATE alembic_version SET version_num='e96a0b8071c';
-- Running upgrade e96a0b8071c -> c6d929b23a8
CREATE TABLE ps_subscription_persistence (
id VARCHAR(40) NOT NULL,
packet VARCHAR(2048),
src_name VARCHAR(128),
src_port INTEGER,
transport_key VARCHAR(64),
local_name VARCHAR(128),
local_port INTEGER,
cseq INTEGER,
tag VARCHAR(128),
endpoint VARCHAR(40),
expires INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id);
UPDATE alembic_version SET version_num='c6d929b23a8';
-- Running upgrade c6d929b23a8 -> 51f8cb66540e
ALTER TABLE ps_endpoints ADD COLUMN force_avp ENUM('yes','no');
ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport ENUM('yes','no');
UPDATE alembic_version SET version_num='51f8cb66540e';
-- Running upgrade 51f8cb66540e -> 1d50859ed02e
ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20);
UPDATE alembic_version SET version_num='1d50859ed02e';
-- Running upgrade 1d50859ed02e -> 1758e8bbf6b
ALTER TABLE sippeers CHANGE useragent useragent VARCHAR(255) NULL;
UPDATE alembic_version SET version_num='1758e8bbf6b';
-- Running upgrade 1758e8bbf6b -> 5139253c0423
ALTER TABLE queue_members DROP COLUMN uniqueid;
ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL;
ALTER TABLE queue_members ADD UNIQUE (uniqueid);
ALTER TABLE queue_members CHANGE uniqueid uniqueid INTEGER NOT NULL AUTO_INCREMENT;
UPDATE alembic_version SET version_num='5139253c0423';
-- Running upgrade 5139253c0423 -> d39508cb8d8
CREATE TABLE queue_rules (
rule_name VARCHAR(80) NOT NULL,
time VARCHAR(32) NOT NULL,
min_penalty VARCHAR(32) NOT NULL,
max_penalty VARCHAR(32) NOT NULL
);
UPDATE alembic_version SET version_num='d39508cb8d8';
-- Running upgrade d39508cb8d8 -> 5950038a6ead
ALTER TABLE ps_transports CHANGE verifiy_server verify_server ENUM('yes','no') NULL;
UPDATE alembic_version SET version_num='5950038a6ead';
-- Running upgrade 5950038a6ead -> 10aedae86a32
ALTER TABLE sippeers CHANGE directmedia directmedia ENUM('yes','no','nonat','update','outgoing') NULL;
UPDATE alembic_version SET version_num='10aedae86a32';
-- Running upgrade 10aedae86a32 -> eb88a14f2a
ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic ENUM('yes','no');
UPDATE alembic_version SET version_num='eb88a14f2a';
-- Running upgrade eb88a14f2a -> 371a3bf4143e
ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone ENUM('yes','no');
UPDATE alembic_version SET version_num='371a3bf4143e';
-- Running upgrade 371a3bf4143e -> 45e3f47c6c44
ALTER TABLE ps_globals ADD COLUMN endpoint_identifier_order VARCHAR(40);
UPDATE alembic_version SET version_num='45e3f47c6c44';
-- Running upgrade 45e3f47c6c44 -> 23530d604b96
ALTER TABLE ps_endpoints ADD COLUMN rpid_immediate ENUM('yes','no');
UPDATE alembic_version SET version_num='23530d604b96';
-- Running upgrade 23530d604b96 -> 31cd4f4891ec
ALTER TABLE ps_endpoints CHANGE dtmf_mode dtmf_mode ENUM('rfc4733','inband','info','auto') NULL;
UPDATE alembic_version SET version_num='31cd4f4891ec';
-- Running upgrade 31cd4f4891ec -> 461d7d691209
ALTER TABLE ps_aors ADD COLUMN qualify_timeout INTEGER;
ALTER TABLE ps_contacts ADD COLUMN qualify_timeout INTEGER;
UPDATE alembic_version SET version_num='461d7d691209';
-- Running upgrade 461d7d691209 -> a541e0b5e89
ALTER TABLE ps_globals ADD COLUMN max_initial_qualify_time INTEGER;
UPDATE alembic_version SET version_num='a541e0b5e89';
-- Running upgrade a541e0b5e89 -> 28b8e71e541f
ALTER TABLE ps_endpoints ADD COLUMN g726_non_standard ENUM('yes','no');
UPDATE alembic_version SET version_num='28b8e71e541f';
-- Running upgrade 28b8e71e541f -> 498357a710ae
ALTER TABLE ps_endpoints ADD COLUMN rtp_keepalive INTEGER;
UPDATE alembic_version SET version_num='498357a710ae';
-- Running upgrade 498357a710ae -> 26f10cadc157
ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout INTEGER;
ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout_hold INTEGER;
UPDATE alembic_version SET version_num='26f10cadc157';
-- Running upgrade 26f10cadc157 -> 154177371065
ALTER TABLE ps_globals ADD COLUMN default_from_user VARCHAR(80);
UPDATE alembic_version SET version_num='154177371065';

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@@ -0,0 +1,34 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 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 CHANGE recording recording BLOB(4294967295) NULL;
UPDATE alembic_version SET version_num='39428242f7f5';

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@@ -0,0 +1,46 @@
SET TRANSACTION READ WRITE
/
CREATE TABLE alembic_version (
version_num VARCHAR2(32 CHAR) NOT NULL
)
/
-- Running upgrade None -> 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')
/
COMMIT
/

File diff suppressed because it is too large Load Diff

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@@ -0,0 +1,52 @@
SET TRANSACTION READ WRITE
/
CREATE TABLE alembic_version (
version_num VARCHAR2(32 CHAR) NOT NULL
)
/
-- Running upgrade None -> 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)
/
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB
/
INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5')
/
COMMIT
/

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@@ -0,0 +1,36 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 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;

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@@ -0,0 +1,787 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 4da0c5f79a9c
CREATE TYPE type_values AS ENUM ('friend','user','peer');
CREATE TYPE sip_transport_values AS ENUM ('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp');
CREATE TYPE sip_dtmfmode_values AS ENUM ('rfc2833','info','shortinfo','inband','auto');
CREATE TYPE sip_directmedia_values AS ENUM ('yes','no','nonat','update');
CREATE TYPE yes_no_values AS ENUM ('yes','no');
CREATE TYPE sip_progressinband_values AS ENUM ('yes','no','never');
CREATE TYPE sip_session_timers_values AS ENUM ('accept','refuse','originate');
CREATE TYPE sip_session_refresher_values AS ENUM ('uac','uas');
CREATE TYPE sip_callingpres_values AS ENUM ('allowed_not_screened','allowed_passed_screen','allowed_failed_screen','allowed','prohib_not_screened','prohib_passed_screen','prohib_failed_screen','prohib');
CREATE TABLE sippeers (
id SERIAL NOT NULL,
name VARCHAR(40) NOT NULL,
ipaddr VARCHAR(45),
port INTEGER,
regseconds INTEGER,
defaultuser VARCHAR(40),
fullcontact VARCHAR(80),
regserver VARCHAR(20),
useragent VARCHAR(20),
lastms INTEGER,
host VARCHAR(40),
type type_values,
context VARCHAR(40),
permit VARCHAR(95),
deny VARCHAR(95),
secret VARCHAR(40),
md5secret VARCHAR(40),
remotesecret VARCHAR(40),
transport sip_transport_values,
dtmfmode sip_dtmfmode_values,
directmedia sip_directmedia_values,
nat VARCHAR(29),
callgroup VARCHAR(40),
pickupgroup VARCHAR(40),
language VARCHAR(40),
disallow VARCHAR(200),
allow VARCHAR(200),
insecure VARCHAR(40),
trustrpid yes_no_values,
progressinband sip_progressinband_values,
promiscredir yes_no_values,
useclientcode yes_no_values,
accountcode VARCHAR(40),
setvar VARCHAR(200),
callerid VARCHAR(40),
amaflags VARCHAR(40),
callcounter yes_no_values,
busylevel INTEGER,
allowoverlap yes_no_values,
allowsubscribe yes_no_values,
videosupport yes_no_values,
maxcallbitrate INTEGER,
rfc2833compensate yes_no_values,
mailbox VARCHAR(40),
"session-timers" sip_session_timers_values,
"session-expires" INTEGER,
"session-minse" INTEGER,
"session-refresher" sip_session_refresher_values,
t38pt_usertpsource VARCHAR(40),
regexten VARCHAR(40),
fromdomain VARCHAR(40),
fromuser VARCHAR(40),
qualify VARCHAR(40),
defaultip VARCHAR(45),
rtptimeout INTEGER,
rtpholdtimeout INTEGER,
sendrpid yes_no_values,
outboundproxy VARCHAR(40),
callbackextension VARCHAR(40),
timert1 INTEGER,
timerb INTEGER,
qualifyfreq INTEGER,
constantssrc yes_no_values,
contactpermit VARCHAR(95),
contactdeny VARCHAR(95),
usereqphone yes_no_values,
textsupport yes_no_values,
faxdetect yes_no_values,
buggymwi yes_no_values,
auth VARCHAR(40),
fullname VARCHAR(40),
trunkname VARCHAR(40),
cid_number VARCHAR(40),
callingpres sip_callingpres_values,
mohinterpret VARCHAR(40),
mohsuggest VARCHAR(40),
parkinglot VARCHAR(40),
hasvoicemail yes_no_values,
subscribemwi yes_no_values,
vmexten VARCHAR(40),
autoframing yes_no_values,
rtpkeepalive INTEGER,
"call-limit" INTEGER,
g726nonstandard yes_no_values,
ignoresdpversion yes_no_values,
allowtransfer yes_no_values,
dynamic yes_no_values,
path VARCHAR(256),
supportpath yes_no_values,
PRIMARY KEY (id),
UNIQUE (name)
);
CREATE INDEX sippeers_name ON sippeers (name);
CREATE INDEX sippeers_name_host ON sippeers (name, host);
CREATE INDEX sippeers_ipaddr_port ON sippeers (ipaddr, port);
CREATE INDEX sippeers_host_port ON sippeers (host, port);
CREATE TYPE iax_requirecalltoken_values AS ENUM ('yes','no','auto');
CREATE TYPE iax_encryption_values AS ENUM ('yes','no','aes128');
CREATE TYPE iax_transfer_values AS ENUM ('yes','no','mediaonly');
CREATE TABLE iaxfriends (
id SERIAL NOT NULL,
name VARCHAR(40) NOT NULL,
type type_values,
username VARCHAR(40),
mailbox VARCHAR(40),
secret VARCHAR(40),
dbsecret VARCHAR(40),
context VARCHAR(40),
regcontext VARCHAR(40),
host VARCHAR(40),
ipaddr VARCHAR(40),
port INTEGER,
defaultip VARCHAR(20),
sourceaddress VARCHAR(20),
mask VARCHAR(20),
regexten VARCHAR(40),
regseconds INTEGER,
accountcode VARCHAR(20),
mohinterpret VARCHAR(20),
mohsuggest VARCHAR(20),
inkeys VARCHAR(40),
outkeys VARCHAR(40),
language VARCHAR(10),
callerid VARCHAR(100),
cid_number VARCHAR(40),
sendani yes_no_values,
fullname VARCHAR(40),
trunk yes_no_values,
auth VARCHAR(20),
maxauthreq INTEGER,
requirecalltoken iax_requirecalltoken_values,
encryption iax_encryption_values,
transfer iax_transfer_values,
jitterbuffer yes_no_values,
forcejitterbuffer yes_no_values,
disallow VARCHAR(200),
allow VARCHAR(200),
codecpriority VARCHAR(40),
qualify VARCHAR(10),
qualifysmoothing yes_no_values,
qualifyfreqok VARCHAR(10),
qualifyfreqnotok VARCHAR(10),
timezone VARCHAR(20),
adsi yes_no_values,
amaflags VARCHAR(20),
setvar VARCHAR(200),
PRIMARY KEY (id),
UNIQUE (name)
);
CREATE INDEX iaxfriends_name ON iaxfriends (name);
CREATE INDEX iaxfriends_name_host ON iaxfriends (name, host);
CREATE INDEX iaxfriends_name_ipaddr_port ON iaxfriends (name, ipaddr, port);
CREATE INDEX iaxfriends_ipaddr_port ON iaxfriends (ipaddr, port);
CREATE INDEX iaxfriends_host_port ON iaxfriends (host, port);
CREATE TABLE voicemail (
uniqueid SERIAL NOT NULL,
context VARCHAR(80) NOT NULL,
mailbox VARCHAR(80) NOT NULL,
password VARCHAR(80) NOT NULL,
fullname VARCHAR(80),
alias VARCHAR(80),
email VARCHAR(80),
pager VARCHAR(80),
attach yes_no_values,
attachfmt VARCHAR(10),
serveremail VARCHAR(80),
language VARCHAR(20),
tz VARCHAR(30),
deletevoicemail yes_no_values,
saycid yes_no_values,
sendvoicemail yes_no_values,
review yes_no_values,
tempgreetwarn yes_no_values,
operator yes_no_values,
envelope yes_no_values,
sayduration INTEGER,
forcename yes_no_values,
forcegreetings yes_no_values,
callback VARCHAR(80),
dialout VARCHAR(80),
exitcontext VARCHAR(80),
maxmsg INTEGER,
volgain NUMERIC(5, 2),
imapuser VARCHAR(80),
imappassword VARCHAR(80),
imapserver VARCHAR(80),
imapport VARCHAR(8),
imapflags VARCHAR(80),
stamp TIMESTAMP WITHOUT TIME ZONE,
PRIMARY KEY (uniqueid)
);
CREATE INDEX voicemail_mailbox ON voicemail (mailbox);
CREATE INDEX voicemail_context ON voicemail (context);
CREATE INDEX voicemail_mailbox_context ON voicemail (mailbox, context);
CREATE INDEX voicemail_imapuser ON voicemail (imapuser);
CREATE TABLE meetme (
bookid SERIAL NOT NULL,
confno VARCHAR(80) NOT NULL,
starttime TIMESTAMP WITHOUT TIME ZONE,
endtime TIMESTAMP WITHOUT TIME ZONE,
pin VARCHAR(20),
adminpin VARCHAR(20),
opts VARCHAR(20),
adminopts VARCHAR(20),
recordingfilename VARCHAR(80),
recordingformat VARCHAR(10),
maxusers INTEGER,
members INTEGER NOT NULL,
PRIMARY KEY (bookid)
);
CREATE INDEX meetme_confno_start_end ON meetme (confno, starttime, endtime);
CREATE TYPE moh_mode_values AS ENUM ('custom','files','mp3nb','quietmp3nb','quietmp3');
CREATE TABLE musiconhold (
name VARCHAR(80) NOT NULL,
mode moh_mode_values,
directory VARCHAR(255),
application VARCHAR(255),
digit VARCHAR(1),
sort VARCHAR(10),
format VARCHAR(10),
stamp TIMESTAMP WITHOUT TIME ZONE,
PRIMARY KEY (name)
);
-- Running upgrade 4da0c5f79a9c -> 43956d550a44
CREATE TYPE yesno_values AS ENUM ('yes','no');
CREATE TYPE pjsip_connected_line_method_values AS ENUM ('invite','reinvite','update');
CREATE TYPE pjsip_direct_media_glare_mitigation_values AS ENUM ('none','outgoing','incoming');
CREATE TYPE pjsip_dtmf_mode_values AS ENUM ('rfc4733','inband','info');
CREATE TYPE pjsip_identify_by_values AS ENUM ('username');
CREATE TYPE pjsip_timer_values AS ENUM ('forced','no','required','yes');
CREATE TYPE pjsip_cid_privacy_values AS ENUM ('allowed_not_screened','allowed_passed_screened','allowed_failed_screened','allowed','prohib_not_screened','prohib_passed_screened','prohib_failed_screened','prohib','unavailable');
CREATE TYPE pjsip_100rel_values AS ENUM ('no','required','yes');
CREATE TYPE pjsip_media_encryption_values AS ENUM ('no','sdes','dtls');
CREATE TYPE pjsip_t38udptl_ec_values AS ENUM ('none','fec','redundancy');
CREATE TYPE pjsip_dtls_setup_values AS ENUM ('active','passive','actpass');
CREATE TABLE ps_endpoints (
id VARCHAR(40) NOT NULL,
transport VARCHAR(40),
aors VARCHAR(200),
auth VARCHAR(40),
context VARCHAR(40),
disallow VARCHAR(200),
allow VARCHAR(200),
direct_media yesno_values,
connected_line_method pjsip_connected_line_method_values,
direct_media_method pjsip_connected_line_method_values,
direct_media_glare_mitigation pjsip_direct_media_glare_mitigation_values,
disable_direct_media_on_nat yesno_values,
dtmf_mode pjsip_dtmf_mode_values,
external_media_address VARCHAR(40),
force_rport yesno_values,
ice_support yesno_values,
identify_by pjsip_identify_by_values,
mailboxes VARCHAR(40),
moh_suggest VARCHAR(40),
outbound_auth VARCHAR(40),
outbound_proxy VARCHAR(40),
rewrite_contact yesno_values,
rtp_ipv6 yesno_values,
rtp_symmetric yesno_values,
send_diversion yesno_values,
send_pai yesno_values,
send_rpid yesno_values,
timers_min_se INTEGER,
timers pjsip_timer_values,
timers_sess_expires INTEGER,
callerid VARCHAR(40),
callerid_privacy pjsip_cid_privacy_values,
callerid_tag VARCHAR(40),
"100rel" pjsip_100rel_values,
aggregate_mwi yesno_values,
trust_id_inbound yesno_values,
trust_id_outbound yesno_values,
use_ptime yesno_values,
use_avpf yesno_values,
media_encryption pjsip_media_encryption_values,
inband_progress yesno_values,
call_group VARCHAR(40),
pickup_group VARCHAR(40),
named_call_group VARCHAR(40),
named_pickup_group VARCHAR(40),
device_state_busy_at INTEGER,
fax_detect yesno_values,
t38_udptl yesno_values,
t38_udptl_ec pjsip_t38udptl_ec_values,
t38_udptl_maxdatagram INTEGER,
t38_udptl_nat yesno_values,
t38_udptl_ipv6 yesno_values,
tone_zone VARCHAR(40),
language VARCHAR(40),
one_touch_recording yesno_values,
record_on_feature VARCHAR(40),
record_off_feature VARCHAR(40),
rtp_engine VARCHAR(40),
allow_transfer yesno_values,
allow_subscribe yesno_values,
sdp_owner VARCHAR(40),
sdp_session VARCHAR(40),
tos_audio INTEGER,
tos_video INTEGER,
cos_audio INTEGER,
cos_video INTEGER,
sub_min_expiry INTEGER,
from_domain VARCHAR(40),
from_user VARCHAR(40),
mwi_fromuser VARCHAR(40),
dtls_verify VARCHAR(40),
dtls_rekey VARCHAR(40),
dtls_cert_file VARCHAR(200),
dtls_private_key VARCHAR(200),
dtls_cipher VARCHAR(200),
dtls_ca_file VARCHAR(200),
dtls_ca_path VARCHAR(200),
dtls_setup pjsip_dtls_setup_values,
srtp_tag_32 yesno_values,
UNIQUE (id)
);
CREATE INDEX ps_endpoints_id ON ps_endpoints (id);
CREATE TYPE pjsip_auth_type_values AS ENUM ('md5','userpass');
CREATE TABLE ps_auths (
id VARCHAR(40) NOT NULL,
auth_type pjsip_auth_type_values,
nonce_lifetime INTEGER,
md5_cred VARCHAR(40),
password VARCHAR(80),
realm VARCHAR(40),
username VARCHAR(40),
UNIQUE (id)
);
CREATE INDEX ps_auths_id ON ps_auths (id);
CREATE TABLE ps_aors (
id VARCHAR(40) NOT NULL,
contact VARCHAR(40),
default_expiration INTEGER,
mailboxes VARCHAR(80),
max_contacts INTEGER,
minimum_expiration INTEGER,
remove_existing yesno_values,
qualify_frequency INTEGER,
authenticate_qualify yesno_values,
UNIQUE (id)
);
CREATE INDEX ps_aors_id ON ps_aors (id);
CREATE TABLE ps_contacts (
id VARCHAR(40) NOT NULL,
uri VARCHAR(40),
expiration_time VARCHAR(40),
qualify_frequency INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_contacts_id ON ps_contacts (id);
CREATE TABLE ps_domain_aliases (
id VARCHAR(40) NOT NULL,
domain VARCHAR(80),
UNIQUE (id)
);
CREATE INDEX ps_domain_aliases_id ON ps_domain_aliases (id);
CREATE TABLE ps_endpoint_id_ips (
id VARCHAR(40) NOT NULL,
endpoint VARCHAR(40),
match VARCHAR(80),
UNIQUE (id)
);
CREATE INDEX ps_endpoint_id_ips_id ON ps_endpoint_id_ips (id);
-- Running upgrade 43956d550a44 -> 581a4264e537
CREATE TABLE extensions (
id BIGSERIAL NOT NULL,
context VARCHAR(40) NOT NULL,
exten VARCHAR(40) NOT NULL,
priority INTEGER NOT NULL,
app VARCHAR(40) NOT NULL,
appdata VARCHAR(256) NOT NULL,
PRIMARY KEY (id, context, exten, priority),
UNIQUE (id)
);
-- Running upgrade 581a4264e537 -> 2fc7930b41b3
CREATE TYPE pjsip_redirect_method_values AS ENUM ('user','uri_core','uri_pjsip');
CREATE TABLE ps_systems (
id VARCHAR(40) NOT NULL,
timer_t1 INTEGER,
timer_b INTEGER,
compact_headers yesno_values,
threadpool_initial_size INTEGER,
threadpool_auto_increment INTEGER,
threadpool_idle_timeout INTEGER,
threadpool_max_size INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_systems_id ON ps_systems (id);
CREATE TABLE ps_globals (
id VARCHAR(40) NOT NULL,
max_forwards INTEGER,
user_agent VARCHAR(40),
default_outbound_endpoint VARCHAR(40),
UNIQUE (id)
);
CREATE INDEX ps_globals_id ON ps_globals (id);
CREATE TYPE pjsip_transport_method_values AS ENUM ('default','unspecified','tlsv1','sslv2','sslv3','sslv23');
CREATE TYPE pjsip_transport_protocol_values AS ENUM ('udp','tcp','tls','ws','wss');
CREATE TABLE ps_transports (
id VARCHAR(40) NOT NULL,
async_operations INTEGER,
bind VARCHAR(40),
ca_list_file VARCHAR(200),
cert_file VARCHAR(200),
cipher VARCHAR(200),
domain VARCHAR(40),
external_media_address VARCHAR(40),
external_signaling_address VARCHAR(40),
external_signaling_port INTEGER,
method pjsip_transport_method_values,
local_net VARCHAR(40),
password VARCHAR(40),
priv_key_file VARCHAR(200),
protocol pjsip_transport_protocol_values,
require_client_cert yesno_values,
verify_client yesno_values,
verifiy_server yesno_values,
tos yesno_values,
cos yesno_values,
UNIQUE (id)
);
CREATE INDEX ps_transports_id ON ps_transports (id);
CREATE TABLE ps_registrations (
id VARCHAR(40) NOT NULL,
auth_rejection_permanent yesno_values,
client_uri VARCHAR(40),
contact_user VARCHAR(40),
expiration INTEGER,
max_retries INTEGER,
outbound_auth VARCHAR(40),
outbound_proxy VARCHAR(40),
retry_interval INTEGER,
forbidden_retry_interval INTEGER,
server_uri VARCHAR(40),
transport VARCHAR(40),
support_path yesno_values,
UNIQUE (id)
);
CREATE INDEX ps_registrations_id ON ps_registrations (id);
ALTER TABLE ps_endpoints ADD COLUMN media_address VARCHAR(40);
ALTER TABLE ps_endpoints ADD COLUMN redirect_method pjsip_redirect_method_values;
ALTER TABLE ps_endpoints ADD COLUMN set_var TEXT;
ALTER TABLE ps_endpoints RENAME mwi_fromuser TO mwi_from_user;
ALTER TABLE ps_contacts ADD COLUMN outbound_proxy VARCHAR(40);
ALTER TABLE ps_contacts ADD COLUMN path TEXT;
ALTER TABLE ps_aors ADD COLUMN maximum_expiration INTEGER;
ALTER TABLE ps_aors ADD COLUMN outbound_proxy VARCHAR(40);
ALTER TABLE ps_aors ADD COLUMN support_path yesno_values;
-- Running upgrade 2fc7930b41b3 -> 21e526ad3040
ALTER TABLE ps_globals ADD COLUMN debug VARCHAR(40);
-- Running upgrade 21e526ad3040 -> 28887f25a46f
CREATE TYPE queue_autopause_values AS ENUM ('yes','no','all');
CREATE TYPE queue_strategy_values AS ENUM ('ringall','leastrecent','fewestcalls','random','rrmemory','linear','wrandom','rrordered');
CREATE TABLE queues (
name VARCHAR(128) NOT NULL,
musiconhold VARCHAR(128),
announce VARCHAR(128),
context VARCHAR(128),
timeout INTEGER,
ringinuse yesno_values,
setinterfacevar yesno_values,
setqueuevar yesno_values,
setqueueentryvar yesno_values,
monitor_format VARCHAR(8),
membermacro VARCHAR(512),
membergosub VARCHAR(512),
queue_youarenext VARCHAR(128),
queue_thereare VARCHAR(128),
queue_callswaiting VARCHAR(128),
queue_quantity1 VARCHAR(128),
queue_quantity2 VARCHAR(128),
queue_holdtime VARCHAR(128),
queue_minutes VARCHAR(128),
queue_minute VARCHAR(128),
queue_seconds VARCHAR(128),
queue_thankyou VARCHAR(128),
queue_callerannounce VARCHAR(128),
queue_reporthold VARCHAR(128),
announce_frequency INTEGER,
announce_to_first_user yesno_values,
min_announce_frequency INTEGER,
announce_round_seconds INTEGER,
announce_holdtime VARCHAR(128),
announce_position VARCHAR(128),
announce_position_limit INTEGER,
periodic_announce VARCHAR(50),
periodic_announce_frequency INTEGER,
relative_periodic_announce yesno_values,
random_periodic_announce yesno_values,
retry INTEGER,
wrapuptime INTEGER,
penaltymemberslimit INTEGER,
autofill yesno_values,
monitor_type VARCHAR(128),
autopause queue_autopause_values,
autopausedelay INTEGER,
autopausebusy yesno_values,
autopauseunavail yesno_values,
maxlen INTEGER,
servicelevel INTEGER,
strategy queue_strategy_values,
joinempty VARCHAR(128),
leavewhenempty VARCHAR(128),
reportholdtime yesno_values,
memberdelay INTEGER,
weight INTEGER,
timeoutrestart yesno_values,
defaultrule VARCHAR(128),
timeoutpriority VARCHAR(128),
PRIMARY KEY (name)
);
CREATE TABLE queue_members (
queue_name VARCHAR(80) NOT NULL,
interface VARCHAR(80) NOT NULL,
uniqueid VARCHAR(80) NOT NULL,
membername VARCHAR(80),
state_interface VARCHAR(80),
penalty INTEGER,
paused INTEGER,
PRIMARY KEY (queue_name, interface)
);
-- Running upgrade 28887f25a46f -> 4c573e7135bd
ALTER TABLE ps_endpoints ALTER COLUMN tos_audio TYPE VARCHAR(10);
ALTER TABLE ps_endpoints ALTER COLUMN tos_video TYPE VARCHAR(10);
ALTER TABLE ps_transports ALTER COLUMN tos TYPE VARCHAR(10);
ALTER TABLE ps_endpoints DROP COLUMN cos_audio;
ALTER TABLE ps_endpoints DROP COLUMN cos_video;
ALTER TABLE ps_transports DROP COLUMN cos;
ALTER TABLE ps_endpoints ADD COLUMN cos_audio INTEGER;
ALTER TABLE ps_endpoints ADD COLUMN cos_video INTEGER;
ALTER TABLE ps_transports ADD COLUMN cos INTEGER;
-- Running upgrade 4c573e7135bd -> 3855ee4e5f85
ALTER TABLE ps_endpoints ADD COLUMN message_context VARCHAR(40);
ALTER TABLE ps_contacts ADD COLUMN user_agent VARCHAR(40);
-- Running upgrade 3855ee4e5f85 -> e96a0b8071c
ALTER TABLE ps_globals ALTER COLUMN user_agent TYPE VARCHAR(255);
ALTER TABLE ps_contacts ALTER COLUMN id TYPE VARCHAR(255);
ALTER TABLE ps_contacts ALTER COLUMN uri TYPE VARCHAR(255);
ALTER TABLE ps_contacts ALTER COLUMN user_agent TYPE VARCHAR(255);
ALTER TABLE ps_registrations ALTER COLUMN client_uri TYPE VARCHAR(255);
ALTER TABLE ps_registrations ALTER COLUMN server_uri TYPE VARCHAR(255);
-- Running upgrade e96a0b8071c -> c6d929b23a8
CREATE TABLE ps_subscription_persistence (
id VARCHAR(40) NOT NULL,
packet VARCHAR(2048),
src_name VARCHAR(128),
src_port INTEGER,
transport_key VARCHAR(64),
local_name VARCHAR(128),
local_port INTEGER,
cseq INTEGER,
tag VARCHAR(128),
endpoint VARCHAR(40),
expires INTEGER,
UNIQUE (id)
);
CREATE INDEX ps_subscription_persistence_id ON ps_subscription_persistence (id);
-- Running upgrade c6d929b23a8 -> 51f8cb66540e
ALTER TABLE ps_endpoints ADD COLUMN force_avp yesno_values;
ALTER TABLE ps_endpoints ADD COLUMN media_use_received_transport yesno_values;
-- Running upgrade 51f8cb66540e -> 1d50859ed02e
ALTER TABLE ps_endpoints ADD COLUMN accountcode VARCHAR(20);
-- Running upgrade 1d50859ed02e -> 1758e8bbf6b
ALTER TABLE sippeers ALTER COLUMN useragent TYPE VARCHAR(255);
-- Running upgrade 1758e8bbf6b -> 5139253c0423
ALTER TABLE queue_members DROP COLUMN uniqueid;
ALTER TABLE queue_members ADD COLUMN uniqueid INTEGER NOT NULL;
ALTER TABLE queue_members ADD UNIQUE (uniqueid);
-- Running upgrade 5139253c0423 -> d39508cb8d8
CREATE TABLE queue_rules (
rule_name VARCHAR(80) NOT NULL,
time VARCHAR(32) NOT NULL,
min_penalty VARCHAR(32) NOT NULL,
max_penalty VARCHAR(32) NOT NULL
);
-- Running upgrade d39508cb8d8 -> 5950038a6ead
ALTER TABLE ps_transports ALTER COLUMN verifiy_server TYPE yesno_values;
ALTER TABLE ps_transports RENAME verifiy_server TO verify_server;
-- Running upgrade 5950038a6ead -> 10aedae86a32
CREATE TYPE sip_directmedia_values_v2 AS ENUM ('yes','no','nonat','update','outgoing');
ALTER TABLE sippeers ALTER COLUMN directmedia TYPE sip_directmedia_values_v2 USING directmedia::text::sip_directmedia_values_v2;
DROP TYPE sip_directmedia_values;
-- Running upgrade 10aedae86a32 -> eb88a14f2a
ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values;
-- Running upgrade eb88a14f2a -> 371a3bf4143e
ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone yesno_values;
-- Running upgrade 371a3bf4143e -> 45e3f47c6c44
ALTER TABLE ps_globals ADD COLUMN endpoint_identifier_order VARCHAR(40);
-- Running upgrade 45e3f47c6c44 -> 23530d604b96
ALTER TABLE ps_endpoints ADD COLUMN rpid_immediate yesno_values;
-- Running upgrade 23530d604b96 -> 31cd4f4891ec
CREATE TYPE pjsip_dtmf_mode_values_v2 AS ENUM ('rfc4733','inband','info','auto');
ALTER TABLE ps_endpoints ALTER COLUMN dtmf_mode TYPE pjsip_dtmf_mode_values_v2 USING dtmf_mode::text::pjsip_dtmf_mode_values_v2;
DROP TYPE pjsip_dtmf_mode_values;
-- Running upgrade 31cd4f4891ec -> 461d7d691209
ALTER TABLE ps_aors ADD COLUMN qualify_timeout INTEGER;
ALTER TABLE ps_contacts ADD COLUMN qualify_timeout INTEGER;
-- Running upgrade 461d7d691209 -> a541e0b5e89
ALTER TABLE ps_globals ADD COLUMN max_initial_qualify_time INTEGER;
-- Running upgrade a541e0b5e89 -> 28b8e71e541f
ALTER TABLE ps_endpoints ADD COLUMN g726_non_standard yesno_values;
-- Running upgrade 28b8e71e541f -> 498357a710ae
ALTER TABLE ps_endpoints ADD COLUMN rtp_keepalive INTEGER;
-- Running upgrade 498357a710ae -> 26f10cadc157
ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout INTEGER;
ALTER TABLE ps_endpoints ADD COLUMN rtp_timeout_hold INTEGER;
-- Running upgrade 26f10cadc157 -> 154177371065
ALTER TABLE ps_globals ADD COLUMN default_from_user VARCHAR(80);
INSERT INTO alembic_version (version_num) VALUES ('154177371065');
COMMIT;

View File

@@ -0,0 +1,36 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade None -> 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);
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
INSERT INTO alembic_version (version_num) VALUES ('39428242f7f5');
COMMIT;

View File

@@ -799,7 +799,7 @@ static void ast_rtp_ice_set_role(struct ast_rtp_instance *instance, enum ast_rtp
role == AST_RTP_ICE_ROLE_CONTROLLED ? "CONTROLLED" : "CONTROLLING", instance);
if (!rtp->ice) {
ast_log(LOG_WARNING, "Set role failed; no ice instance (%p)\n", instance);
ast_debug(3, "Set role failed; no ice instance (%p)\n", instance);
return;
}
@@ -3199,8 +3199,8 @@ static int ast_rtcp_write(const void *data)
/*
* Not being rescheduled.
*/
ao2_ref(instance, -1);
rtp->rtcp->schedid = -1;
ao2_ref(instance, -1);
}
return res;
@@ -3311,7 +3311,7 @@ static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame
rtp->txcount++;
rtp->txoctetcount += (res - hdrlen);
if (rtp->rtcp && rtp->rtcp->schedid < 1) {
if (rtp->rtcp && rtp->rtcp->schedid < 0) {
ast_debug(1, "Starting RTCP transmission on RTP instance '%p'\n", instance);
ao2_ref(instance, +1);
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);
@@ -4522,7 +4522,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
}
/* Do not schedule RR if RTCP isn't run */
if (rtp->rtcp && !ast_sockaddr_isnull(&rtp->rtcp->them) && rtp->rtcp->schedid < 1) {
if (rtp->rtcp && !ast_sockaddr_isnull(&rtp->rtcp->them) && rtp->rtcp->schedid < 0) {
/* Schedule transmission of Receiver Report */
ao2_ref(instance, +1);
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);