kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-02 19:16:15 +00:00
Code Issues Packages Projects Releases Wiki Activity

Update for 13.17.0-rc1

Browse Source
This commit is contained in:
George Joseph
2017-07-06 06:52:04 -05:00
parent 5a894ff27e
commit 0c00ee754b
17 changed files with 59262 additions and 0 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
.lastclean Normal file
View File

@@ -0,0 +1 @@
40

1
.version Normal file
View File

@@ -0,0 +1 @@
13.17.0-rc1

52059
ChangeLog Normal file
View File

File diff suppressed because it is too large Load Diff

311
asterisk-13.17.0-rc1-summary.html Normal file
View File

@@ -0,0 +1,311 @@
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-13.17.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-13.17.0-rc1</h3><h3 align="center">Date: 2017-07-06</h3><h3 align="center">&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-13.16.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">17 Sean Bright <sean.bright@gmail.com><br/>11 George Joseph <gjoseph@digium.com><br/>10 Joshua Colp <jcolp@digium.com><br/>9 Alexei Gradinari <alex2grad@gmail.com><br/>5 Richard Mudgett <rmudgett@digium.com><br/>5 Kevin Harwell <kharwell@digium.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>1 Jan Friesse <jfriesse@redhat.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/>1 David M. Lee <dlee@digium.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Rodrigo Ramirez Norambuena <a@rodrigoramirez.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 Corey Farrell <git@cfware.com><br/></td><td width="33%"><td width="33%">4 Alexei Gradinari <alex2grad@gmail.com><br/>4 Joshua Colp <jcolp@digium.com><br/>3 Kevin Harwell <kharwell@digium.com><br/>3 Louis Jocelyn Paquet <ljpaquet@quebecinternet.net><br/>3 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>3 George Joseph <gjoseph@digium.com><br/>2 Guido Falsi <madpilot@freebsd.org><br/>2 Alexander Traud <pabstraud@compuserve.com><br/>2 Michael Walton <mike@farsouthnet.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>1 Rusty Newton <rnewton@digium.com><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Jacek Konieczny <jkonieczny@eggsoft.pl><br/>1 Tim Morgan <morganuci@gmail.com><br/>1 Etienne Allovon <eallovon@avencall.com><br/>1 alex <asterisk@maximum.guru><br/>1 Kinsey Moore <kmoore@digium.com><br/>1 John Harris <john.harris@certus-tech.com><br/>1 Javier Riveros <goseeped@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Robert Mordec <r.mordec@slican.pl><br/>1 Ross Beer <ross.beer@voicehost.co.uk><br/>1 Chris Howard <choward@digium.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Andrew Nowrot <andrew.nowrot@gmail.com><br/>1 'alex'<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Ben Langfeld <ben@langfeld.me><br/>1 John Fawcett <john@voipsupport.it><br/>1 Corey Farrell <git@cfware.com><br/>1 Frankie Chin <fchin@biamp.com><br/>1 Zach R <zrothy@monmouth.com><br/>1 Matthias Binder <it@mitterhuemer.at><br/>1 Christopher van de Sande <cvandesande@opendmz.com><br/>1 Stefan Engström <stefanen@kth.se><br/>1 Antoine Pitrou <pitrou@free.fr><br/>1 Alex <metsys@gmx.com><br/>1 Etienne Lessard <elessard97@gmail.com><br/>1 Ryan Smith <ryan.smith@tekara.co.uk><br/>1 Michael Maier <m1278468@mailbox.org><br/>1 OpenBSD ports<br/>1 Marek Cervenka <marek.cervenka@gmail.com><br/>1 Ronald Raikes <reraikes@avweb.com><br/>1 Ove Aursand <oveaurs@gmail.com><br/>1 Richard Mudgett <rmudgett@digium.com><br/>1 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>1 wushumasters <wushumasters@gmail.com><br/>1 Tony Mountifield <tony@softins.co.uk><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Michel R. Vaillancourt <michel@jkl5group.com><br/>1 David Brillert <david_brillert@scopserv.com><br/>1 Yasin CANER <yasin.caner@netgsm.com.tr><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Addons/format_mp3</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23951">ASTERISK-23951</a>: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97b003f5e2d4a350508fc20173e180a23f8ef525">[97b003f5e2]</a> Sean Bright -- format_mp3: Re-work menuselect/build issues</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72213c98e3d4d5287ed321f1b4fb67087a7a129c">[72213c98e3]</a> Sean Bright -- format_mp3: Don't try to build format_mp3 if we don't have sources</li>
</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27012">ASTERISK-27012</a>: app_confbridge: ConfBridge sometimes does not play user name recording while leaving<br/>Reported by: Robert Mordec<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1b32de2c5fb8854183f0c7d8c9df7470ab9c140">[f1b32de2c5]</a> Robert Mordec -- app_confbridge: Race between removing and playing name recording while leaving</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25665">ASTERISK-25665</a>: Duplicate logging in queue log for EXITEMPTY events<br/>Reported by: Ove Aursand<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26400">ASTERISK-26400</a>: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26715">ASTERISK-26715</a>: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel<br/>Reported by: David Brillert<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26975">ASTERISK-26975</a>: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call<br/>Reported by: Lorne Gaetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24052">ASTERISK-24052</a>: app_voicemail reloads result in leaked IMAP sockets.<br/>Reported by: Louis Jocelyn Paquet<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b6c327c515944d74aa798f385e01768a4bb04c2">[3b6c327c51]</a> Alexei Gradinari -- app_voicemail: IMAP logout on reload/unload</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08be5e01e8ab72a7e9e80525e20967467a6df99b">[08be5e01e8]</a> Alexei Gradinari -- app_voicemail: IMAP logout on MWI unsubscribe</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27039">ASTERISK-27039</a>: chan_pjsip: Device state is idle when channel from endpoint is in early media<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f10c6b3b044f9979c523f65f449670047dcb57f">[1f10c6b3b0]</a> Joshua Colp -- chan_pjsip: Update device state when in early media.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26996">ASTERISK-26996</a>: chan_pjsip: Flipping between codecs<br/>Reported by: Michael Maier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=996a4791ff123e80d71d44cb0fd13bb201d197b1">[996a4791ff]</a> Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec' option to include us changing.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26281">ASTERISK-26281</a>: chan_pjsip would send INVITE to 'Unreachable' endpoints<br/>Reported by: Jacek Konieczny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=746c2c574578608a6b48d4794ba33cda5a6dd484">[746c2c5745]</a> Joshua Colp -- res_pjsip: Add support for returning only reachable contacts and use it.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4479038073e57a67c19c1ec5dc8896fcc8c3a0fb">[4479038073]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
</ul><br><h4>Category: Channels/chan_sip/SRTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27075">ASTERISK-27075</a>: bridge: stuck channel(s) after failed attended transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=67664fbf95a00ced30f8791fd1089b4595e29479">[67664fbf95]</a> Kevin Harwell -- bridge: stuck channel(s) after failed attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27074">ASTERISK-27074</a>: core_local: local channel data not being properly unref'ed and unlocked<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f9913f2723cbcbf6d78f4da7ee4dd4decc13c05">[1f9913f272]</a> Kevin Harwell -- core_local: local channel data not being properly unref'ed and unlocked</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26923">ASTERISK-26923</a>: bridging: T.38 request is lost when channels are added to bridge<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e414833f6e77345f4969116972e9cf1ad9b595fd">[e414833f6e]</a> Joshua Colp -- bridge: Add a deferred queue.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27025">ASTERISK-27025</a>: channel / meetme: Fix missing parentheses<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc05183f4b7d728534ec6fa5f3fc21802396aabf">[dc05183f4b]</a> Joshua Colp -- channel / app_meetme: Fix parentheses.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26789">ASTERISK-26789</a>: Audit manipulation of channel flags without locks<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=161820396495a549c9a378d32136cbb5f28ef2af">[1618203964]</a> Joshua Colp -- asterisk: Audit locking of channel when manipulating flags.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27041">ASTERISK-27041</a>: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc307af7f2ed653914aeadb0b7e613cb4e239b06">[dc307af7f2]</a> Frederic LE FOLL -- Core/PBX: Deadlock between dialplan execution and application unregistration.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27057">ASTERISK-27057</a>: Seg Fault in ast_sorcery_object_get_id at sorcery.c<br/>Reported by: Ryan Smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2eea791e4178e5f2e4446a5f70d81ac27cf2a0e">[c2eea791e4]</a> George Joseph -- res_pjsip_pubsub: Fix reference to released endpoint</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839">ASTERISK-23839</a>: AGI - RECORD FILE - documentation doesn't describe BEEP argument<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3eb7fbba72482b3019a7493c68e533e67d9d8235">[3eb7fbba72]</a> Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27060">ASTERISK-27060</a>: Comment typo format_g729.c<br/>Reported by: Matthew Fredrickson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a40073750b46ae28ddf1041d5ed3ab57151298e">[0a40073750]</a> Matthew Fredrickson -- formats/format_g729: Fix typo in comment</li>
</ul><br><h4>Category: PBX/pbx_realtime</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19291">ASTERISK-19291</a>: Background in realtime<br/>Reported by: Andrew Nowrot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=283cc59af746896a2b2bc23899fc86118895f7c0">[283cc59af7]</a> Sean Bright -- pbx_builtin: Properly handle hangup during Background</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23839">ASTERISK-23839</a>: AGI - RECORD FILE - documentation doesn't describe BEEP argument<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3eb7fbba72482b3019a7493c68e533e67d9d8235">[3eb7fbba72]</a> Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22432">ASTERISK-22432</a>: Async AGI crashes Asterisk when issuing "set variable" command without args<br/>Reported by: Antoine Pitrou<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f306e451f6f905a2bb74c15cb844735c244a7610">[f306e451f6]</a> Sean Bright -- res_agi: Prevent crash when SET VARIABLE called without arguments</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25662">ASTERISK-25662</a>: Malformed AGI 520 Usage response<br/>Reported by: Tony Mountifield<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a007e438c36960d4179e2f188767e7ae14a204d1">[a007e438c3]</a> Sean Bright -- res_agi: Fix malformed AGI usage response</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27026">ASTERISK-27026</a>: res_ari: Crash when no ari.conf configuration file exists<br/>Reported by: Ronald Raikes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7901b9853e8f60e1d2dce44ce81dec6f7f866ccc">[7901b9853e]</a> George Joseph -- res_ari: Add "module loaded" check to ari stubs</li>
</ul><br><h4>Category: Resources/res_ari_recordings</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27021">ASTERISK-27021</a>: GET /recordings/stored returns 500 Internal Server Error<br/>Reported by: Tim Morgan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf6cf59646f52dc3de12dac16c3c3824ce9ae927">[cf6cf59646]</a> Sean Bright -- stasis_recording: Correct ast_asprintf error checking</li>
</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27008">ASTERISK-27008</a>: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space<br/>Reported by: John Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=700ef6861ab966008ca16e5f23c64eb68b047c08">[700ef6861a]</a> Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp attributes</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26399">ASTERISK-26399</a>: app_queue: Agent not called when caller is parked<br/>Reported by: wushumasters<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bfcb1acc7ae53d50e1b784b4d46c588744aae8b">[6bfcb1acc7]</a> Joshua Colp -- app_queue: Fix members showing as being in call when not.</li>
</ul><br><h4>Category: Resources/res_pjsip/Bundling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27052">ASTERISK-27052</a>: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network<br/>Reported by: alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0bde568669ac26735c1058115ae96223a7e69a6b">[0bde568669]</a> George Joseph -- pjproject_bundled: Use the asterisk github mirror for download</li>
</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053">ASTERISK-27053</a>: res_pjsip_refer/session: Calls dropped during transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27024">ASTERISK-27024</a>: nat/external_media settings ignored in 14.4.1<br/>Reported by: Christopher van de Sande<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2dee95cc7a280d0ab84c778bf44a76aa62ac758b">[2dee95cc7a]</a> Florian Floimair -- res_pjsip_session: Correct inverted test in session_outgoing_nat_hook</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27053">ASTERISK-27053</a>: res_pjsip_refer/session: Calls dropped during transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cdf3191d3538b2e9a1aec31580db1e01d73d5ef">[6cdf3191d3]</a> Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26964">ASTERISK-26964</a>: res_pjsip_session: Wrong From on reinvite when request and To URI differ<br/>Reported by: Yasin CANER<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36628cc9c474b52b134a415803b14f87e420dce6">[36628cc9c4]</a> Yasin CANER -- res_pjsip_session : fixed wrong From Header number On Re-invite</li>
</ul><br><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27046">ASTERISK-27046</a>: res_pjsip_transport_websocket: segfault in get_write_timeout<br/>Reported by: Jørgen H<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e16a669c70c5a93bb9a38c218a5348cd62bd780a">[e16a669c70]</a> Jørgen H -- res_pjsip_transport_websocket: Add NULL check in get_write_timeout</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27022">ASTERISK-27022</a>: res_rtp_asterisk: Incorrect SSRC change for RTCP component<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7dafe82751fd512d58bb3843601daff013958dd2">[7dafe82751]</a> George Joseph -- res_rtp_asterisk: Fix ssrc change for rtcp srtp</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24858">ASTERISK-24858</a>: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec<br/>Reported by: Frankie Chin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=70e5887906db8d585892409cde89e5e28111549a">[70e5887906]</a> Sean Bright -- format: Reintroduce smoother flags</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26982">ASTERISK-26982</a>: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4479038073e57a67c19c1ec5dc8896fcc8c3a0fb">[4479038073]</a> Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX</li>
</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25294">ASTERISK-25294</a>: srtp's crypto_get_random deprecated<br/>Reported by: Tzafrir Cohen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25101">ASTERISK-25101</a>: DTLS configuration can not be specified in the general section - documentation<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=971a401ce95ed0f566b2e90a52d69d0274c63ff8">[971a401ce9]</a> Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26979">ASTERISK-26979</a>: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e91efef2bb35cd0b03f45ad1b1ba43203948368d">[e91efef2bb]</a> Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm</li>
</ul><br><h4>Category: Resources/res_stasis_snoop</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26973">ASTERISK-26973</a>: bridge: Crash when freeing frame and snooping<br/>Reported by: Michel R. Vaillancourt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26333">ASTERISK-26333</a>: Problems with Blind Transfer, PJSIP (Aastra 6869i)<br/>Reported by: Matthias Binder<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6af2dd34afc2c20bdabd07bc3836821690db4c86">[6af2dd34af]</a> Alexei Gradinari -- res_pjsip: New endpoint option "refer_blind_progress"</li>
</ul><br><h3>Information Request</h3><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26976">ASTERISK-26976</a>: libsrtp-2.x.x support<br/>Reported by: Alex<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e9cd1f20d86de1c25b7a9accffb7d3e2601878b">[5e9cd1f20d]</a> Sean Bright -- res_srtp: Add support for libsrtp2</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27043">ASTERISK-27043</a>: Core/BuildSystem: Add defines to fix build with LibreSSL<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a64f65fe6fee96702668bdd3344233f19232850">[6a64f65fe6]</a> Guido Falsi -- BuildSystem: Add patches to allow building with recent LibreSSL</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419">ASTERISK-26419</a>: audiohooks: Remove redundant codec translations when using audiohooks<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26419">ASTERISK-26419</a>: audiohooks: Remove redundant codec translations when using audiohooks<br/>Reported by: Michael Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=adfb28882bfd2055d8b54705805db573d8ce6c94">[adfb28882b]</a> Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks</li>
</ul><br><h4>Category: Core/Portability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27042">ASTERISK-27042</a>: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44cee2f4a15db911d2c9bdd6f845d17a1e6c6c17">[44cee2f4a1]</a> Guido Falsi -- BuildSystem: Fix build on FreeBSD due to missing crypt.h</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26124">ASTERISK-26124</a>: res_agi: Set audio format for EAGI audio stream<br/>Reported by: John Fawcett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90237dca11d0adf129198cef4a6a0716a52618b5">[90237dca11]</a> Sean Bright -- res_agi: Allow configuration of audio format of EAGI pipe</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26230">ASTERISK-26230</a>: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f6a9617eb44a8d59b5828cd860d3852cc824ce9">[0f6a9617eb]</a> Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59c9bbe6961a5677ddb13eed2a130d16b6ffc0ee">[59c9bbe696]</a> Alexei Gradinari -- res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled</li>
</ul><br><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Bug</h3><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27065">ASTERISK-27065</a>: call hangup after leaving app_queue<br/>Reported by: Marek Cervenka<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c43ca0ac50764ab17d691844a84158bbf590b0e">[2c43ca0ac5]</a> Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26469">ASTERISK-26469</a>: Infinite loop after a dual Redirect<br/>Reported by: Etienne Allovon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b07b2162359ccc9a3f84324fabce18b6ad63eee3">[b07b216235]</a> Joshua Colp -- manager: Clear the flag on the other channel.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27095">ASTERISK-27095</a>: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bd7c0f37cb7b513d1333717ece0118bd8875546">[6bd7c0f37c]</a> George Joseph -- chan_pjsip: Fix ability to send UPDATE on COLP</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27106">ASTERISK-27106</a>: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS.<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39d2ebbf56635355432eb96ff850c0c9bf2a5d63">[39d2ebbf56]</a> Alexander Traud -- chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support).</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9f4b3b966e911fae157a484d8f4a1440130eede6">[9f4b3b966e]</a> Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support).</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27016">ASTERISK-27016</a>: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.<br/>Reported by: Chris Howard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4910a3bf402baddf8ed72badfaed7ae64da48686">[4910a3bf40]</a> Joshua Colp -- channel: Fix reference counting in ast_channel_suppress.</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27100">ASTERISK-27100</a>: channel: ast_waitfordigit_full fails to clear flag in an error branch.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73520e9f58857049a086fb88106e342cdc25d3a1">[73520e9f58]</a> Corey Farrell -- channel: Clear channel flag in error branch.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26978">ASTERISK-26978</a>: rtp: Crash in ast_rtp_codecs_payload_code()<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb48e99bd4f4556424a6799e2e5f7aebf8911e8d">[eb48e99bd4]</a> George Joseph -- bridge_native_rtp: Keep rtp instance refs on bridge_channel</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27108">ASTERISK-27108</a>: Crash using 'data get' CLI command<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6258de458b2e6ba02e91ed67bbd2801f0984526a">[6258de458b]</a> Sean Bright -- core: Fix segfault when invoking 'data get' CLI command</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27088">ASTERISK-27088</a>: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0426b1d88ab97c4fc1b2b27f8da93b28096f2dfc">[0426b1d88a]</a> Joshua Colp -- res_rtp_asterisk: Fix issues with ICE renegotiation.</li>
</ul><br><h4>Category: Resources/res_corosync</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25370">ASTERISK-25370</a>: res_corosync segfaults at startup with corosync version > 2.x<br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=005a4afa6b0e710e11b47b11cfc152b028c596fc">[005a4afa6b]</a> Jan Friesse -- res_corosync: Change thread stack size</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27090">ASTERISK-27090</a>: PJSIP: Deadlock using TCP transport<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d64cbde5756eaa1c7ee62116e112b7ebd198bbe">[0d64cbde57]</a> Richard Mudgett -- pjsip_distributor.c: Fix deadlock with TCP type transports.</li>
</ul><br><h4>Category: Resources/res_pjsip_dialog_info_body_generator</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26919">ASTERISK-26919</a>: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip<br/>Reported by: Zach R<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6e4899612ca71bc3c9180dadea0c0117e8ae462">[a6e4899612]</a> Alexei Gradinari -- res_pjsip: New endpoint option "notify_early_inuse_ringing"</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27051">ASTERISK-27051</a>: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e749c8f51c20fb13bfe93e969cf02d7e74cdb27">[8e749c8f51]</a> Alexei Gradinari -- res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27059">ASTERISK-27059</a>: res_stasis: Stolen channel references are leaking<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=edfdb4dff5d8438bdb1dfb526c57618944ea6bf3">[edfdb4dff5]</a> George Joseph -- res_stasis: Plug reference leak on stolen channels</li>
</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27097">ASTERISK-27097</a>: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbe68f139db525b2d922f63d8452d9732fb5f1b9">[bbe68f139d]</a> George Joseph -- pjproject_bundled: Allow passing configure options to bundled</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27068">ASTERISK-27068</a>: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f356192d196ae146b0c2390f8d62024694e691f">[8f356192d1]</a> Alexei Gradinari -- app_voicemail: IMAP connection control</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fbc34d2bd5393d93d8b3b3a8c6daa895c2e9633">[9fbc34d2bd]</a> Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27066">ASTERISK-27066</a>: res_pjsip: Add DTMF INFO Failback mode<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fbc34d2bd5393d93d8b3b3a8c6daa895c2e9633">[9fbc34d2bd]</a> Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode</li>
</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=379fe658312e11699ff8c8e8a463e31b3c277237">379fe65831</a></td><td>George Joseph</td><td>Fix alembic branches</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=905d18e8bf52ea7657acaaf2ec0cbe58531fb625">905d18e8bf</a></td><td>Richard Mudgett</td><td>pjsip_distributor.c: Fix unidentified_requests hash functions.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f59d08924bc676970cabc6f3e291c7d1d2f2707">1f59d08924</a></td><td>Torrey Searle</td><td>res/res_pjsip_t38: fix incorrect increment of media_count</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=764d04fa8705d9e5c2e7aee8a6f1c774d7d28595">764d04fa87</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Eliminate RAII_VAR in contact delete observer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cecf6540dc4779598289f711340bb966bbfcc6aa">cecf6540dc</a></td><td>Rodrigo Ramírez Norambuena</td><td>cdr: fix mistake spelling of a word for Unanswered.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9a4ab8c8c00c8d53584d6f7e31729b5027c8dd6">b9a4ab8c8c</a></td><td>Richard Mudgett</td><td>chan_pjsip: Fix PJSIP_MEDIA_OFFER dialplan function read.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1a209d5ac8f8b7fe96e54d6aba55dbf0dbb1403">f1a209d5ac</a></td><td>Richard Mudgett</td><td>app_voicemail.c: Fix compile error when IMAP enabled.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68de35a6a01e2a1fe732e156b73f800bb672a421">68de35a6a0</a></td><td>David M. Lee</td><td>CFLAGS for BIND8 support</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da3312457e6cf1c0d7bc8cb2a4aba57877fb5afc">da3312457e</a></td><td>Sean Bright</td><td>codecs.conf.sample: Fix max_bandwidth speling error</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=590ffcaf0b03bbe3d25730ad750a2075a46c7208">590ffcaf0b</a></td><td>Sean Bright</td><td>eventfd: Disable during cross compilation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5520b6c201875133a73db5a2c88b5fc5b78864bb">5520b6c201</a></td><td>Alexei Gradinari</td><td>CHANGES: correct version for a new option 'refer_blind_progress'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c093bf8072ff65bf29d290c1330291c460cd7fdf">c093bf8072</a></td><td>Sean Bright</td><td>res_rtp_multicast: Use consistent timestamps when possible</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c10341646d353922b4ee92c77fc4e5560d263c73">c10341646d</a></td><td>George Joseph</td><td>test_json: Fix test names with reserved words</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=65898c3af82e2d780a48d9d50d3b1c952c208a89">65898c3af8</a></td><td>George Joseph</td><td>unittests: Add a unit test that causes a SEGV and...</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>.lastclean | 1
.version | 1
ChangeLog |51038 ----------
asterisk-13.16.0-summary.html | 405
asterisk-13.16.0-summary.txt | 952
b/CHANGES | 54
b/Makefile | 3
b/addons/Makefile | 10
b/apps/app_chanspy.c | 16
b/apps/app_confbridge.c | 79
b/apps/app_dial.c | 6
b/apps/app_disa.c | 10
b/apps/app_dumpchan.c | 4
b/apps/app_externalivr.c | 6
b/apps/app_meetme.c | 2
b/apps/app_queue.c | 109
b/apps/app_voicemail.c | 80
b/autoconf/ast_ext_lib.m4 | 36
b/bridges/bridge_native_rtp.c | 677
b/bridges/bridge_simple.c | 32
b/channels/chan_pjsip.c | 68
b/channels/chan_sip.c | 8
b/channels/pjsip/dialplan_functions.c | 37
b/configs/samples/cdr.conf.sample | 2
b/configs/samples/codecs.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 20
b/configs/samples/sip.conf.sample | 3
b/configs/samples/voicemail.conf.sample | 3
b/configure | 434
b/configure.ac | 100
b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
b/formats/format_g729.c | 2
b/include/asterisk/ari.h | 10
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/bridge_channel.h | 2
b/include/asterisk/bridge_channel_internal.h | 11
b/include/asterisk/bridge_technology.h | 3
b/include/asterisk/channel.h | 25
b/include/asterisk/codec.h | 3
b/include/asterisk/core_local.h | 37
b/include/asterisk/format.h | 11
b/include/asterisk/res_pjsip.h | 74
b/include/asterisk/res_pjsip_presence_xml.h | 3
b/include/asterisk/res_pjsip_session.h | 11
b/include/asterisk/rtp_engine.h | 9
b/include/asterisk/smoother.h | 1
b/include/asterisk/test.h | 8
b/main/autoservice.c | 2
b/main/bridge.c | 10
b/main/bridge_after.c | 2
b/main/bridge_channel.c | 38
b/main/channel.c | 90
b/main/codec_builtin.c | 19
b/main/core_local.c | 54
b/main/crypt.c | 2
b/main/data.c | 4
b/main/file.c | 20
b/main/format.c | 8
b/main/libasteriskssl.c | 4
b/main/manager.c | 8
b/main/pbx.c | 4
b/main/pbx_app.c | 7
b/main/pbx_builtins.c | 8
b/main/tcptls.c | 4
b/main/test.c | 4
b/makeopts.in | 2
b/res/res_agi.c | 73
b/res/res_ari_applications.c | 4
b/res/res_ari_asterisk.c | 4
b/res/res_ari_bridges.c | 4
b/res/res_ari_channels.c | 4
b/res/res_ari_device_states.c | 4
b/res/res_ari_endpoints.c | 4
b/res/res_ari_events.c | 33
b/res/res_ari_mailboxes.c | 4
b/res/res_ari_playbacks.c | 4
b/res/res_ari_recordings.c | 4
b/res/res_ari_sounds.c | 4
b/res/res_corosync.c | 29
b/res/res_format_attr_h263.c | 2
b/res/res_format_attr_h264.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip.c | 31
b/res/res_pjsip/location.c | 53
b/res/res_pjsip/pjsip_configuration.c | 9
b/res/res_pjsip/pjsip_distributor.c | 242
b/res/res_pjsip/presence_xml.c | 9
b/res/res_pjsip_dialog_info_body_generator.c | 10
b/res/res_pjsip_mwi.c | 87
b/res/res_pjsip_pidf_body_generator.c | 2
b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 8
b/res/res_pjsip_refer.c | 28
b/res/res_pjsip_sdp_rtp.c | 38
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_session.exports.in | 1
b/res/res_pjsip_t38.c | 2
b/res/res_pjsip_transport_websocket.c | 4
b/res/res_pjsip_xpidf_body_generator.c | 2
b/res/res_rtp_asterisk.c | 41
b/res/res_rtp_multicast.c | 139
b/res/res_srtp.c | 15
b/res/res_stasis.c | 20
b/res/srtp/srtp_compat.h | 29
b/res/stasis_recording/stored.c | 4
b/rest-api-templates/res_ari_resource.c.mustache | 35
b/tests/test_bridging.c | 292
b/tests/test_json.c | 16
b/tests/test_pbx.c | 22
b/third-party/configure.m4 | 5
b/third-party/pjproject/Makefile | 2
b/third-party/pjproject/Makefile.rules | 7
b/third-party/pjproject/configure.m4 | 6
contrib/realtime/mssql/mssql_cdr.sql | 44
contrib/realtime/mssql/mssql_config.sql | 1713
contrib/realtime/mssql/mssql_voicemail.sql | 54
contrib/realtime/mysql/mysql_cdr.sql | 32
contrib/realtime/mysql/mysql_config.sql | 1052
contrib/realtime/mysql/mysql_voicemail.sql | 34
contrib/realtime/oracle/oracle_cdr.sql | 38
contrib/realtime/oracle/oracle_config.sql | 1707
contrib/realtime/oracle/oracle_voicemail.sql | 48
contrib/realtime/postgresql/postgresql_cdr.sql | 36
contrib/realtime/postgresql/postgresql_config.sql | 1130
contrib/realtime/postgresql/postgresql_voicemail.sql | 38
127 files changed, 3137 insertions(+), 58993 deletions(-)</pre><br></html>

832
asterisk-13.17.0-rc1-summary.txt Normal file
View File

@@ -0,0 +1,832 @@
Release Summary
asterisk-13.17.0-rc1
Date: 2017-07-06
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Open Issues
5. Other Changes
6. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This release is a point release of an existing major version. The changes
included were made to address problems that have been identified in this
release series, or are minor, backwards compatible new features or
improvements. Users should be able to safely upgrade to this version if
this release series is already in use. Users considering upgrading from a
previous version are strongly encouraged to review the UPGRADE.txt
document as well as the CHANGES document for information about upgrading
to this release series.
The data in this summary reflects changes that have been made since the
previous release, asterisk-13.16.0.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were affected by commits that went into
this release.
Coders Testers Reporters
17 Sean Bright 4 Alexei Gradinari
11 George Joseph 4 Joshua Colp
10 Joshua Colp 3 Kevin Harwell
9 Alexei Gradinari 3 Louis Jocelyn Paquet
5 Richard Mudgett 3 Tzafrir Cohen
5 Kevin Harwell 3 George Joseph
2 Torrey Searle 2 Guido Falsi
2 Guido Falsi 2 Alexander Traud
2 Alexander Traud 2 Michael Walton
1 Jan Friesse 2 Torrey Searle
1 Florian Floimair 1 Rusty Newton
1 Ivan Poddubny 1 Matthew Fredrickson
1 Matthew Fredrickson 1 Jacek Konieczny
1 Yasin CANER 1 Tim Morgan
1 David M. Lee 1 Etienne Allovon
1 Robert Mordec 1 alex
1 JA,rgen H 1 Kinsey Moore
1 Rodrigo Ramirez Norambuena 1 John Harris
1 Frederic LE FOLL 1 Javier Riveros
1 Corey Farrell 1 Sean Bright
1 Robert Mordec
1 Ross Beer
1 Chris Howard
1 mdu113
1 Andrew Nowrot
1 'alex'
1 Lorne Gaetz
1 Ben Langfeld
1 John Fawcett
1 Corey Farrell
1 Frankie Chin
1 Zach R
1 Matthias Binder
1 Christopher van de Sande
1 Stefan EngstrAP:m
1 Antoine Pitrou
1 Alex
1 Etienne Lessard
1 Ryan Smith
1 Michael Maier
1 OpenBSD ports
1 Marek Cervenka
1 Ronald Raikes
1 Ove Aursand
1 Richard Mudgett
1 Frederic LE FOLL
1 wushumasters
1 Tony Mountifield
1 JA,rgen H
1 Michel R. Vaillancourt
1 David Brillert
1 Yasin CANER
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Bug
Category: Addons/format_mp3
ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if
mp3lib was not downloaded
Reported by: Tzafrir Cohen
* [97b003f5e2] Sean Bright -- format_mp3: Re-work menuselect/build
issues
* [72213c98e3] Sean Bright -- format_mp3: Don't try to build format_mp3
if we don't have sources
Category: Applications/app_confbridge
ASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user
name recording while leaving
Reported by: Robert Mordec
* [f1b32de2c5] Robert Mordec -- app_confbridge: Race between removing
and playing name recording while leaving
Category: Applications/app_meetme
ASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua Colp
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
Category: Applications/app_queue
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove Aursand
* [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
when a queue is empty
ASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumasters
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26400: app_queue: Queue member stops being called after AMI
"Redirect" action for queues with wrapuptime
Reported by: Etienne Lessard
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26715: app_queue: Member will not receive any new calls after
doing a transfer if wrapuptime = greater than 0 and using Local channel
Reported by: David Brillert
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
ASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to
receive queue calls after transfer queue call
Reported by: Lorne Gaetz
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
Category: Applications/app_voicemail/IMAP
ASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets.
Reported by: Louis Jocelyn Paquet
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
control
* [3b6c327c51] Alexei Gradinari -- app_voicemail: IMAP logout on
reload/unload
* [08be5e01e8] Alexei Gradinari -- app_voicemail: IMAP logout on MWI
unsubscribe
Category: Bridges/bridge_simple
ASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. Vaillancourt
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Channels/chan_pjsip
ASTERISK-27039: chan_pjsip: Device state is idle when channel from
endpoint is in early media
Reported by: Joshua Colp
* [1f10c6b3b0] Joshua Colp -- chan_pjsip: Update device state when in
early media.
ASTERISK-26996: chan_pjsip: Flipping between codecs
Reported by: Michael Maier
* [996a4791ff] Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec'
option to include us changing.
ASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints
Reported by: Jacek Konieczny
* [746c2c5745] Joshua Colp -- res_pjsip: Add support for returning only
reachable contacts and use it.
Category: Channels/chan_sip/General
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngstrAP:m
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
Category: Channels/chan_sip/SRTP
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
Category: Codecs/General
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
Category: Core/Bridging
ASTERISK-27075: bridge: stuck channel(s) after failed attended transfer
Reported by: Kevin Harwell
* [67664fbf95] Kevin Harwell -- bridge: stuck channel(s) after failed
attended transfer
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
bridge
Reported by: Torrey Searle
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
Category: Core/Channels
ASTERISK-27074: core_local: local channel data not being properly unref'ed
and unlocked
Reported by: Kevin Harwell
* [1f9913f272] Kevin Harwell -- core_local: local channel data not being
properly unref'ed and unlocked
ASTERISK-26923: bridging: T.38 request is lost when channels are added to
bridge
Reported by: Torrey Searle
* [e414833f6e] Joshua Colp -- bridge: Add a deferred queue.
ASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua Colp
* [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses.
Category: Core/General
ASTERISK-26789: Audit manipulation of channel flags without locks
Reported by: Joshua Colp
* [1618203964] Joshua Colp -- asterisk: Audit locking of channel when
manipulating flags.
Category: Core/PBX
ASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and
application unregistration
Reported by: Frederic LE FOLL
* [dc307af7f2] Frederic LE FOLL -- Core/PBX: Deadlock between dialplan
execution and application unregistration.
Category: Core/RTP
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
Category: Core/Sorcery
ASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c
Reported by: Ryan Smith
* [c2eea791e4] George Joseph -- res_pjsip_pubsub: Fix reference to
released endpoint
Category: Documentation
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
argument
Reported by: Rusty Newton
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
documentation
Category: General
ASTERISK-27060: Comment typo format_g729.c
Reported by: Matthew Fredrickson
* [0a40073750] Matthew Fredrickson -- formats/format_g729: Fix typo in
comment
Category: PBX/pbx_realtime
ASTERISK-19291: Background in realtime
Reported by: Andrew Nowrot
* [283cc59af7] Sean Bright -- pbx_builtin: Properly handle hangup during
Background
Category: Resources/res_agi
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP
argument
Reported by: Rusty Newton
* [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE'
documentation
ASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable"
command without args
Reported by: Antoine Pitrou
* [f306e451f6] Sean Bright -- res_agi: Prevent crash when SET VARIABLE
called without arguments
ASTERISK-25662: Malformed AGI 520 Usage response
Reported by: Tony Mountifield
* [a007e438c3] Sean Bright -- res_agi: Fix malformed AGI usage response
Category: Resources/res_ari
ASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists
Reported by: Ronald Raikes
* [7901b9853e] George Joseph -- res_ari: Add "module loaded" check to
ari stubs
Category: Resources/res_ari_recordings
ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error
Reported by: Tim Morgan
* [cf6cf59646] Sean Bright -- stasis_recording: Correct ast_asprintf
error checking
Category: Resources/res_format_attr_h264
ASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional
parameters have a space
Reported by: John Harris
* [700ef6861a] Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp
attributes
Category: Resources/res_parking
ASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumasters
* [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in
call when not.
Category: Resources/res_pjsip/Bundling
ASTERISK-27052: Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow network
Reported by: alex
* [0bde568669] George Joseph -- pjproject_bundled: Use the asterisk
github mirror for download
Category: Resources/res_pjsip_refer
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin Harwell
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
during transfer
Category: Resources/res_pjsip_session
ASTERISK-27024: nat/external_media settings ignored in 14.4.1
Reported by: Christopher van de Sande
* [2dee95cc7a] Florian Floimair -- res_pjsip_session: Correct inverted
test in session_outgoing_nat_hook
ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin Harwell
* [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped
during transfer
ASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and
To URI differ
Reported by: Yasin CANER
* [36628cc9c4] Yasin CANER -- res_pjsip_session : fixed wrong From
Header number On Re-invite
Category: Resources/res_pjsip_transport_websocket
ASTERISK-27046: res_pjsip_transport_websocket: segfault in
get_write_timeout
Reported by: JA,rgen H
* [e16a669c70] JA,rgen H -- res_pjsip_transport_websocket: Add NULL
check in get_write_timeout
Category: Resources/res_rtp_asterisk
ASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component
Reported by: Michael Walton
* [7dafe82751] George Joseph -- res_rtp_asterisk: Fix ssrc change for
rtcp srtp
ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte
order on Intel platform when using slin codec
Reported by: Frankie Chin
* [70e5887906] Sean Bright -- format: Reintroduce smoother flags
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
Reported by: Javier Riveros
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
wrong srtp unprotecting algorithm
ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion
failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngstrAP:m
* [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX
Category: Resources/res_srtp
ASTERISK-25294: srtp's crypto_get_random deprecated
Reported by: Tzafrir Cohen
* [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
ASTERISK-25101: DTLS configuration can not be specified in the general
section - documentation
Reported by: Ben Langfeld
* [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS
settings are permitted
ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
Reported by: Javier Riveros
* [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the
wrong srtp unprotecting algorithm
Category: Resources/res_stasis_snoop
ASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. Vaillancourt
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: pjproject/pjsip
ASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i)
Reported by: Matthias Binder
* [6af2dd34af] Alexei Gradinari -- res_pjsip: New endpoint option
"refer_blind_progress"
Information Request
Category: Resources/res_rtp_asterisk
ASTERISK-26976: libsrtp-2.x.x support
Reported by: Alex
* [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2
Improvement
Category: Core/BuildSystem
ASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL
Reported by: Guido Falsi
* [6a64f65fe6] Guido Falsi -- BuildSystem: Add patches to allow building
with recent LibreSSL
Category: Core/Channels
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
audiohooks
Reported by: Michael Walton
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Core/General
ASTERISK-26419: audiohooks: Remove redundant codec translations when using
audiohooks
Reported by: Michael Walton
* [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed
after call to audiohooks
Category: Core/Portability
ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to
missing crypt.h file
Reported by: Guido Falsi
* [44cee2f4a1] Guido Falsi -- BuildSystem: Fix build on FreeBSD due to
missing crypt.h
Category: Resources/res_agi
ASTERISK-26124: res_agi: Set audio format for EAGI audio stream
Reported by: John Fawcett
* [90237dca11] Sean Bright -- res_agi: Allow configuration of audio
format of EAGI pipe
Category: Resources/res_pjsip_mwi
ASTERISK-26230: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP
taskprocessor on startup
Reported by: Alexei Gradinari
* [0f6a9617eb] Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI
subscriptions on updating contact
* [59c9bbe696] Alexei Gradinari -- res_pjsip_mwi: don't create mwi
subscriptions if initial unsolicited disabled
----------------------------------------------------------------------
Open Issues
[Back to Top]
This is a list of all open issues from the issue tracker that were
referenced by changes that went into this release.
Bug
Category: Applications/app_queue
ASTERISK-27065: call hangup after leaving app_queue
Reported by: Marek Cervenka
* [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan
when a queue is empty
Category: Bridges/bridge_simple
ASTERISK-26469: Infinite loop after a dual Redirect
Reported by: Etienne Allovon
* [b07b216235] Joshua Colp -- manager: Clear the flag on the other
channel.
Category: Channels/chan_pjsip
ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite,
we're not trying UPDATE
Reported by: George Joseph
* [6bd7c0f37c] George Joseph -- chan_pjsip: Fix ability to send UPDATE
on COLP
Category: Channels/chan_sip/General
ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really
different domain with TLS.
Reported by: Alexander Traud
* [39d2ebbf56] Alexander Traud -- chan_sip: Only when different, add
TCP|TLS in autodomain (SIP Domain Support).
* [9f4b3b966e] Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr
in autodomain (SIP Domain Support).
Category: Core/Bridging
ASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events'
bridge is muted multiple times.
Reported by: Chris Howard
* [4910a3bf40] Joshua Colp -- channel: Fix reference counting in
ast_channel_suppress.
Category: Core/Channels
ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an
error branch.
Reported by: Corey Farrell
* [73520e9f58] Corey Farrell -- channel: Clear channel flag in error
branch.
Category: Core/RTP
ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code()
Reported by: Ross Beer
* [eb48e99bd4] George Joseph -- bridge_native_rtp: Keep rtp instance
refs on bridge_channel
Category: General
ASTERISK-27108: Crash using 'data get' CLI command
Reported by: Sean Bright
* [6258de458b] Sean Bright -- core: Fix segfault when invoking 'data
get' CLI command
ASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and
unidirectional negotiation
Reported by: Joshua Colp
* [0426b1d88a] Joshua Colp -- res_rtp_asterisk: Fix issues with ICE
renegotiation.
Category: Resources/res_corosync
ASTERISK-25370: res_corosync segfaults at startup with corosync version >
2.x
Reported by: mdu113
* [005a4afa6b] Jan Friesse -- res_corosync: Change thread stack size
Category: Resources/res_pjsip
ASTERISK-27090: PJSIP: Deadlock using TCP transport
Reported by: Richard Mudgett
* [0d64cbde57] Richard Mudgett -- pjsip_distributor.c: Fix deadlock with
TCP type transports.
Category: Resources/res_pjsip_dialog_info_body_generator
ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse
behavior difference between chan_sip and res_pjsip
Reported by: Zach R
* [a6e4899612] Alexei Gradinari -- res_pjsip: New endpoint option
"notify_early_inuse_ringing"
Category: Resources/res_pjsip_mwi
ASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on
deleting the endpoint's last contact
Reported by: Alexei Gradinari
* [8e749c8f51] Alexei Gradinari -- res_pjsip_mwi: unsubscribe
unsolicited MWI on deleting endpoint last contact
Category: Resources/res_stasis
ASTERISK-27059: res_stasis: Stolen channel references are leaking
Reported by: George Joseph
* [edfdb4dff5] George Joseph -- res_stasis: Plug reference leak on
stolen channels
Category: Third-Party/pjproject
ASTERISK-27097: pjproject_bundled: We don't pass options needed for
cross-compile to pjproject configure
Reported by: George Joseph
* [bbe68f139d] George Joseph -- pjproject_bundled: Allow passing
configure options to bundled
Improvement
Category: Applications/app_voicemail/IMAP
ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to
specify post-polling disconnect
Reported by: Alexei Gradinari
* [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection
control
Category: Channels/chan_pjsip
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey Searle
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
Category: Resources/res_pjsip
ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey Searle
* [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+------------------+----------------------------------------|
| 379fe65831 | George Joseph | Fix alembic branches |
|------------+------------------+----------------------------------------|
| 905d18e8bf | Richard Mudgett | pjsip_distributor.c: Fix |
| | | unidentified_requests hash functions. |
|------------+------------------+----------------------------------------|
| 1f59d08924 | Torrey Searle | res/res_pjsip_t38: fix incorrect |
| | | increment of media_count |
|------------+------------------+----------------------------------------|
| 764d04fa87 | Richard Mudgett | res_pjsip_mwi.c: Eliminate RAII_VAR in |
| | | contact delete observer |
|------------+------------------+----------------------------------------|
| cecf6540dc | Rodrigo RamArez | cdr: fix mistake spelling of a word |
| | Norambuena | for Unanswered. |
|------------+------------------+----------------------------------------|
| b9a4ab8c8c | Richard Mudgett | chan_pjsip: Fix PJSIP_MEDIA_OFFER |
| | | dialplan function read. |
|------------+------------------+----------------------------------------|
| f1a209d5ac | Richard Mudgett | app_voicemail.c: Fix compile error |
| | | when IMAP enabled. |
|------------+------------------+----------------------------------------|
| 68de35a6a0 | David M. Lee | CFLAGS for BIND8 support |
|------------+------------------+----------------------------------------|
| da3312457e | Sean Bright | codecs.conf.sample: Fix max_bandwidth |
| | | speling error |
|------------+------------------+----------------------------------------|
| 590ffcaf0b | Sean Bright | eventfd: Disable during cross |
| | | compilation |
|------------+------------------+----------------------------------------|
| 5520b6c201 | Alexei Gradinari | CHANGES: correct version for a new |
| | | option 'refer_blind_progress' |
|------------+------------------+----------------------------------------|
| c093bf8072 | Sean Bright | res_rtp_multicast: Use consistent |
| | | timestamps when possible |
|------------+------------------+----------------------------------------|
| c10341646d | George Joseph | test_json: Fix test names with |
| | | reserved words |
|------------+------------------+----------------------------------------|
| 65898c3af8 | George Joseph | unittests: Add a unit test that causes |
| | | a SEGV and... |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
.lastclean | 1
.version | 1
ChangeLog |51038 ----------
asterisk-13.16.0-summary.html | 405
asterisk-13.16.0-summary.txt | 952
b/CHANGES | 54
b/Makefile | 3
b/addons/Makefile | 10
b/apps/app_chanspy.c | 16
b/apps/app_confbridge.c | 79
b/apps/app_dial.c | 6
b/apps/app_disa.c | 10
b/apps/app_dumpchan.c | 4
b/apps/app_externalivr.c | 6
b/apps/app_meetme.c | 2
b/apps/app_queue.c | 109
b/apps/app_voicemail.c | 80
b/autoconf/ast_ext_lib.m4 | 36
b/bridges/bridge_native_rtp.c | 677
b/bridges/bridge_simple.c | 32
b/channels/chan_pjsip.c | 68
b/channels/chan_sip.c | 8
b/channels/pjsip/dialplan_functions.c | 37
b/configs/samples/cdr.conf.sample | 2
b/configs/samples/codecs.conf.sample | 6
b/configs/samples/pjsip.conf.sample | 20
b/configs/samples/sip.conf.sample | 3
b/configs/samples/voicemail.conf.sample | 3
b/configure | 434
b/configure.ac | 100
b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58
b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30
b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30
b/formats/format_g729.c | 2
b/include/asterisk/ari.h | 10
b/include/asterisk/autoconfig.h.in | 3
b/include/asterisk/bridge_channel.h | 2
b/include/asterisk/bridge_channel_internal.h | 11
b/include/asterisk/bridge_technology.h | 3
b/include/asterisk/channel.h | 25
b/include/asterisk/codec.h | 3
b/include/asterisk/core_local.h | 37
b/include/asterisk/format.h | 11
b/include/asterisk/res_pjsip.h | 74
b/include/asterisk/res_pjsip_presence_xml.h | 3
b/include/asterisk/res_pjsip_session.h | 11
b/include/asterisk/rtp_engine.h | 9
b/include/asterisk/smoother.h | 1
b/include/asterisk/test.h | 8
b/main/autoservice.c | 2
b/main/bridge.c | 10
b/main/bridge_after.c | 2
b/main/bridge_channel.c | 38
b/main/channel.c | 90
b/main/codec_builtin.c | 19
b/main/core_local.c | 54
b/main/crypt.c | 2
b/main/data.c | 4
b/main/file.c | 20
b/main/format.c | 8
b/main/libasteriskssl.c | 4
b/main/manager.c | 8
b/main/pbx.c | 4
b/main/pbx_app.c | 7
b/main/pbx_builtins.c | 8
b/main/tcptls.c | 4
b/main/test.c | 4
b/makeopts.in | 2
b/res/res_agi.c | 73
b/res/res_ari_applications.c | 4
b/res/res_ari_asterisk.c | 4
b/res/res_ari_bridges.c | 4
b/res/res_ari_channels.c | 4
b/res/res_ari_device_states.c | 4
b/res/res_ari_endpoints.c | 4
b/res/res_ari_events.c | 33
b/res/res_ari_mailboxes.c | 4
b/res/res_ari_playbacks.c | 4
b/res/res_ari_recordings.c | 4
b/res/res_ari_sounds.c | 4
b/res/res_corosync.c | 29
b/res/res_format_attr_h263.c | 2
b/res/res_format_attr_h264.c | 2
b/res/res_musiconhold.c | 4
b/res/res_pjsip.c | 31
b/res/res_pjsip/location.c | 53
b/res/res_pjsip/pjsip_configuration.c | 9
b/res/res_pjsip/pjsip_distributor.c | 242
b/res/res_pjsip/presence_xml.c | 9
b/res/res_pjsip_dialog_info_body_generator.c | 10
b/res/res_pjsip_mwi.c | 87
b/res/res_pjsip_pidf_body_generator.c | 2
b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2
b/res/res_pjsip_pubsub.c | 8
b/res/res_pjsip_refer.c | 28
b/res/res_pjsip_sdp_rtp.c | 38
b/res/res_pjsip_session.c | 37
b/res/res_pjsip_session.exports.in | 1
b/res/res_pjsip_t38.c | 2
b/res/res_pjsip_transport_websocket.c | 4
b/res/res_pjsip_xpidf_body_generator.c | 2
b/res/res_rtp_asterisk.c | 41
b/res/res_rtp_multicast.c | 139
b/res/res_srtp.c | 15
b/res/res_stasis.c | 20
b/res/srtp/srtp_compat.h | 29
b/res/stasis_recording/stored.c | 4
b/rest-api-templates/res_ari_resource.c.mustache | 35
b/tests/test_bridging.c | 292
b/tests/test_json.c | 16
b/tests/test_pbx.c | 22
b/third-party/configure.m4 | 5
b/third-party/pjproject/Makefile | 2
b/third-party/pjproject/Makefile.rules | 7
b/third-party/pjproject/configure.m4 | 6
contrib/realtime/mssql/mssql_cdr.sql | 44
contrib/realtime/mssql/mssql_config.sql | 1713
contrib/realtime/mssql/mssql_voicemail.sql | 54
contrib/realtime/mysql/mysql_cdr.sql | 32
contrib/realtime/mysql/mysql_config.sql | 1052
contrib/realtime/mysql/mysql_voicemail.sql | 34
contrib/realtime/oracle/oracle_cdr.sql | 38
contrib/realtime/oracle/oracle_config.sql | 1707
contrib/realtime/oracle/oracle_voicemail.sql | 48
contrib/realtime/postgresql/postgresql_cdr.sql | 36
contrib/realtime/postgresql/postgresql_config.sql | 1130
contrib/realtime/postgresql/postgresql_voicemail.sql | 38
127 files changed, 3137 insertions(+), 58993 deletions(-)

44
contrib/realtime/mssql/mssql_cdr.sql Normal file
View File

@@ -0,0 +1,44 @@
BEGIN TRANSACTION;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
GO
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20) NULL,
src VARCHAR(80) NULL,
dst VARCHAR(80) NULL,
dcontext VARCHAR(80) NULL,
clid VARCHAR(80) NULL,
channel VARCHAR(80) NULL,
dstchannel VARCHAR(80) NULL,
lastapp VARCHAR(80) NULL,
lastdata VARCHAR(80) NULL,
start DATETIME NULL,
answer DATETIME NULL,
[end] DATETIME NULL,
duration INTEGER NULL,
billsec INTEGER NULL,
disposition VARCHAR(45) NULL,
amaflags VARCHAR(45) NULL,
userfield VARCHAR(256) NULL,
uniqueid VARCHAR(150) NULL,
linkedid VARCHAR(150) NULL,
peeraccount VARCHAR(20) NULL,
sequence INTEGER NULL
);
GO
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
GO
COMMIT;
GO

1759
contrib/realtime/mssql/mssql_config.sql Normal file
View File

File diff suppressed because it is too large Load Diff

54
contrib/realtime/mssql/mssql_voicemail.sql Normal file
View File

@@ -0,0 +1,54 @@
BEGIN TRANSACTION;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
GO
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80) NULL,
macrocontext VARCHAR(80) NULL,
callerid VARCHAR(80) NULL,
origtime INTEGER NULL,
duration INTEGER NULL,
recording IMAGE NULL,
flag VARCHAR(30) NULL,
category VARCHAR(30) NULL,
mailboxuser VARCHAR(30) NULL,
mailboxcontext VARCHAR(30) NULL,
msg_id VARCHAR(40) NULL
);
GO
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
GO
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
GO
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
GO
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording IMAGE;
GO
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
GO
COMMIT;
GO

32
contrib/realtime/mysql/mysql_cdr.sql Normal file
View File

@@ -0,0 +1,32 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start DATETIME,
answer DATETIME,
end DATETIME,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');

1070
contrib/realtime/mysql/mysql_config.sql Normal file
View File

File diff suppressed because it is too large Load Diff

34
contrib/realtime/mysql/mysql_voicemail.sql Normal file
View File

@@ -0,0 +1,34 @@
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BLOB,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB(4294967295) NULL;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';

38
contrib/realtime/oracle/oracle_cdr.sql Normal file
View File

@@ -0,0 +1,38 @@
CREATE TABLE alembic_version (
version_num VARCHAR2(32 CHAR) NOT NULL
)
/
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR2(20 CHAR),
src VARCHAR2(80 CHAR),
dst VARCHAR2(80 CHAR),
dcontext VARCHAR2(80 CHAR),
clid VARCHAR2(80 CHAR),
channel VARCHAR2(80 CHAR),
dstchannel VARCHAR2(80 CHAR),
lastapp VARCHAR2(80 CHAR),
lastdata VARCHAR2(80 CHAR),
"start" DATE,
answer DATE,
end DATE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR2(45 CHAR),
amaflags VARCHAR2(45 CHAR),
userfield VARCHAR2(256 CHAR),
uniqueid VARCHAR2(150 CHAR),
linkedid VARCHAR2(150 CHAR),
peeraccount VARCHAR2(20 CHAR),
sequence INTEGER
)
/
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d')
/

1753
contrib/realtime/oracle/oracle_config.sql Normal file
View File

File diff suppressed because it is too large Load Diff

48
contrib/realtime/oracle/oracle_voicemail.sql Normal file
View File

@@ -0,0 +1,48 @@
CREATE TABLE alembic_version (
version_num VARCHAR2(32 CHAR) NOT NULL
)
/
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR2(255 CHAR) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR2(80 CHAR),
macrocontext VARCHAR2(80 CHAR),
callerid VARCHAR2(80 CHAR),
origtime INTEGER,
duration INTEGER,
recording BLOB,
flag VARCHAR2(30 CHAR),
category VARCHAR2(30 CHAR),
mailboxuser VARCHAR2(30 CHAR),
mailboxcontext VARCHAR2(30 CHAR),
msg_id VARCHAR2(40 CHAR)
)
/
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum)
/
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir)
/
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e')
/
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages MODIFY recording BLOB
/
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e'
/

36
contrib/realtime/postgresql/postgresql_cdr.sql Normal file
View File

@@ -0,0 +1,36 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade -> 210693f3123d
CREATE TABLE cdr (
accountcode VARCHAR(20),
src VARCHAR(80),
dst VARCHAR(80),
dcontext VARCHAR(80),
clid VARCHAR(80),
channel VARCHAR(80),
dstchannel VARCHAR(80),
lastapp VARCHAR(80),
lastdata VARCHAR(80),
start TIMESTAMP WITHOUT TIME ZONE,
answer TIMESTAMP WITHOUT TIME ZONE,
"end" TIMESTAMP WITHOUT TIME ZONE,
duration INTEGER,
billsec INTEGER,
disposition VARCHAR(45),
amaflags VARCHAR(45),
userfield VARCHAR(256),
uniqueid VARCHAR(150),
linkedid VARCHAR(150),
peeraccount VARCHAR(20),
sequence INTEGER
);
INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
COMMIT;

1152
contrib/realtime/postgresql/postgresql_config.sql Normal file
View File

File diff suppressed because it is too large Load Diff

38
contrib/realtime/postgresql/postgresql_voicemail.sql Normal file
View File

@@ -0,0 +1,38 @@
BEGIN;
CREATE TABLE alembic_version (
version_num VARCHAR(32) NOT NULL
);
-- Running upgrade -> a2e9769475e
CREATE TABLE voicemail_messages (
dir VARCHAR(255) NOT NULL,
msgnum INTEGER NOT NULL,
context VARCHAR(80),
macrocontext VARCHAR(80),
callerid VARCHAR(80),
origtime INTEGER,
duration INTEGER,
recording BYTEA,
flag VARCHAR(30),
category VARCHAR(30),
mailboxuser VARCHAR(30),
mailboxcontext VARCHAR(30),
msg_id VARCHAR(40)
);
ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIMARY KEY (dir, msgnum);
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
-- Running upgrade a2e9769475e -> 39428242f7f5
ALTER TABLE voicemail_messages ALTER COLUMN recording TYPE BYTEA;
UPDATE alembic_version SET version_num='39428242f7f5' WHERE alembic_version.version_num = 'a2e9769475e';
COMMIT;