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Update for 17.1.0-rc2

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Asterisk Development Team
2019-12-18 10:51:49 -05:00
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17.1.0-rc1
17.1.0-rc2

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ChangeLog
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2019-12-18 15:51 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 17.1.0-rc2 Released.
2019-12-18 08:30 +0000 [a76d3103fb] George Joseph <gjoseph@digium.com>
* Revert "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up"
There are reports that this commit causes deadlocks when issuing
a "pri show" CLI command and a channel is in the process of being
hung up. Further work is in progress to determine the cause of
the deadlock and provide a permanent fix.
ASTERISK~28605 #keeping open
Reported by: Dirk Wendland
This reverts commit c6b17b521231dde9da890d95ee705c93953bab8c.
Change-Id: Iddd900c1abdd5074ff39c17cdce855f2f436cef9
2019-12-04 15:01 +0000 [27a4a3c761] George Joseph <gjoseph@digium.com>
* res_rtp_asterisk: Add frame list cleanups to ast_rtp_read
In Asterisk 16+, there are a few places in ast_rtp_read where we've
allocated a frame list but return a null frame instead of the list.
In these cases, any frames left in the list won't be freed. In the
vast majority of the cases, the list is empty when we return so
there's nothing to free but there have been leaks reported in the
wild that can be traced back to frames left in the list before
returning.
The escape paths now all have logic to free frames left in the
list.
ASTERISK-28609
Reported by: Ted G
Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a
2019-12-16 06:35 +0000 [3d29b06e37] Joshua C. Colp <jcolp@sangoma.com>
* configure: Add check for MySQL client bool and my_bool type usage.
Instead of trying to use the defined MySQL client version from the
header use a configure check to determine whether the bool or my_bool
type should be used for defining a boolean.
ASTERISK-28604
Change-Id: Id2225b3785115de074c50c123ff1a68005b4a9c7
2019-12-16 05:23 +0000 [5949f9a86a] Joshua C. Colp <jcolp@sangoma.com>
* res_pjsip_session: Set stream state on created streams for incoming SDP.
A previous review, 13174, made a change whereby on an incoming offer SDP
the pending topology was initialized to the configured. This caused a problem
for bundle with WebRTC where bundle could reference a stream that did not
actually exist if the configuration had both audio and video but the
offer SDP only contained audio.
This change undoes that review and instead fixes the original problem it
sought to solve by setting the state of created streams based on the
contents of the offer SDP. This way the stream state is not inactive
until negotiation later completes.
ASTERISK-28659
Change-Id: Ic5ae5a86437d3e686ac5afd91d133cc916198355
2019-12-12 11:29 +0000 Asterisk Development Team <asteriskteam@digium.com>
* asterisk 17.1.0-rc1 Released.

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asterisk-17.1.0-rc1-summary.html
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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-17.1.0-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.1.0-rc1</h3><h3 align="center">Date: 2019-12-12</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-17.0.0.</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%">23 Sean Bright <sean.bright@gmail.com><br/>21 George Joseph <gjoseph@digium.com><br/>16 Joshua Colp <jcolp@digium.com><br/>13 Kevin Harwell <kharwell@digium.com><br/>6 Corey Farrell <git@cfware.com><br/>5 Alexei Gradinari <alex2grad@gmail.com><br/>5 Ben Ford <bford@digium.com><br/>4 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>3 Asterisk Development Team <asteriskteam@digium.com><br/>3 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>2 Salah Ahmed <txrubel@gmail.com><br/>2 Torrey Searle <torrey@voxbone.com><br/>2 lvl <digium@lvlconsultancy.nl><br/>1 Thomas Arimont (license 5525)<br/>1 Pascal Cadotte Michaud <pcm@wazo.io><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Jonathan Rose <jrose@digium.com><br/>1 Michael Goryainov<br/>1 Chris-Savinovich <csavinovich@digium.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 sungtae kim <pchero21@gmail.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 cmaj <chris@penguinpbx.com><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Guido Falsi <madpilot@FreeBSD.org><br/></td><td width="33%">1 tests/test_utils.c.<br/></td><td width="33%">5 Kevin Harwell <kharwell@digium.com><br/>5 Joshua C. Colp <jcolp@digium.com><br/>4 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>3 Salah Ahmed <txrubel@gmail.com><br/>3 Ross Beer <ross.beer@voicehost.co.uk><br/>2 Alexei Gradinari <alex2grad@gmail.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>2 Ross Beer<br/>2 Joshua Elson <joshelson@gmail.com><br/>2 Ruddy G <plugworld@micnes.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Walter Doekes <walter+asterisk@wjd.nu><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 Byron Clark <bclark@getjive.com><br/>1 Niklas Larsson <niklas@tese.se><br/>1 Jonas Swiatek <jonas@telzio.com><br/>1 Yoooooo Ha <n1906374c@e.ntu.edu.sg><br/>1 Michael <ringo@vianet.ca><br/>1 Eliel Sardañons <eliels@gmail.com><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Dan Cropp<br/>1 Jeremiah Gadd <jeremygadd@gmail.com><br/>1 Bernhard Schmidt<br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Marian Piater <marian.piater@voipsun.cz><br/>1 Pascal Cadotte Michaud <pascal.cadotte@gmail.com><br/>1 Kilburn <kilburna@gmail.com><br/>1 Michael Goryainov <gms4nlt@gmail.com><br/>1 Bernhard Schmidt <berni@birkenwald.de><br/>1 Ian Jones <tech@iljones.net><br/>1 Alexander Traud<br/>1 Aheliotech <phones@aheliotech.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Mark <mark@wrapped.cx><br/>1 Timothy Vanderaerden <timothy.vanderaerden@optimise-group.be><br/>1 Niklas Larsson<br/>1 Andrey V. T. <avt1203@gmail.com><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Speed Dial Dave <speed_dial_dave@gmx.com><br/>1 Jonathan Harris <lardconcepts@gmail.com><br/>1 Daniel <depeee@gmail.com><br/>1 Sam Banks <sam.banks.nz@gmail.com><br/>1 George Joseph <gjoseph@digium.com><br/>1 Eliel Sardañons<br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Cyril Ramière <cyril.ramiere@ino.global><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 cmaj <chris@penguinpbx.com><br/>1 Juan Martin <jmartin79@yandex.com><br/>1 lvl <digium@lvlconsultancy.nl><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>Security</h3><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28589">ASTERISK-28589</a>: chan_sip: Depending on configuration an INVITE can alter Addr of a peer<br/>Reported by: Andrey V. T.<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=665a94cb7622dd5f9591df5d68962e41266ee47b">[665a94cb76]</a> Ben Ford -- chan_sip.c: Prevent address change on unauthenticated SIP request.</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28580">ASTERISK-28580</a>: Bypass SYSTEM write permission in manager action allows system commands execution<br/>Reported by: Eliel Sardañons<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1ba589671a0e6e9a52b89af6e5398f040ef99f">[6b1ba58967]</a> George Joseph -- manager.c: Prevent the Originate action from running the Originate app</li>
</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28495">ASTERISK-28495</a>: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d4f1e8ebe5dc1152c59aee60955fc29ae3b3191">[9d4f1e8ebe]</a> Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28567">ASTERISK-28567</a>: Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.<br/>Reported by: Michael<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b90399498787b898da89d21564251e6aa84dda98">[b903994987]</a> Sean Bright -- Revert "app_voicemail: Cleanup stale lock files on module load"</li>
</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22192">ASTERISK-22192</a>: [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column<br/>Reported by: cmaj<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa0973f868ac463bc31b760f364afb2ab691b030">[aa0973f868]</a> cmaj -- app_voicemail.c: Support multiple file formats for forwarded messages.</li>
</ul><br><h4>Category: Core/CodecInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28512">ASTERISK-28512</a>: Add pass-through support for H.265 (HEVC) codec<br/>Reported by: Florian Floimair<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d7a3e4f5cf6dadc0c8fa9b0cdc0c2d13b45d33ab">[d7a3e4f5cf]</a> Florian Floimair -- core: Add H.265/HEVC passthrough support</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28586">ASTERISK-28586</a>: Typo in README-SERIOUSLY.bestpractices.md<br/>Reported by: Sam Banks<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4bc1c170cd09ef890133e438e7030ac9b6eeba85">[4bc1c170cd]</a> Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling correetions.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28542">ASTERISK-28542</a>: [patch] add the ability for asterisk to generate on-hold re-invites<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=55b760d7625f3c78c8f4de5587a2051b7a16dfa4">[55b760d762]</a> Torrey Searle -- channel/chan_pjsip: add dialplan function for music on hold</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28602">ASTERISK-28602</a>: res_pjsip_outbound_registration: Maximum retries reached<br/>Reported by: Daniel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eea2d499f4cc0982ff8817640c803dbd57a008ed">[eea2d499f4]</a> Joshua Colp -- res_pjsip_outbound_registration: Extend documentation for "max_retries".</li>
</ul><br><h3>Bug</h3><h4>Category: .Release/Targets</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28488">ASTERISK-28488</a>: pjsip mwi: n+1 sip notify's sent on re-register<br/>Reported by: Chris Savinovich<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a36fb473fe3cda2fd56a44f4e09234a167ef44d2">[a36fb473fe]</a> Kevin Harwell -- res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions</li>
</ul><br><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28608">ASTERISK-28608</a>: app_amd: Use time calculation to calculate timeout<br/>Reported by: Michael Cargile<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e23b2856d09a7843e3d8dda6f1f3b72de89a48e6">[e23b2856d0]</a> Michael Cargile -- app_amd: Fixed timeout issue</li>
</ul><br><h4>Category: Applications/app_chanisavail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28527">ASTERISK-28527</a>: ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50997de8878f0ad08147540395dab3bc4ffb0d2a">[50997de887]</a> Frederic LE FOLL -- ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf.</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28604">ASTERISK-28604</a>: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36b28c98dd8fd8906b9b6eefee2489820f1a1295">[36b28c98dd]</a> George Joseph -- Build: Fix compile issues with seldom used modules</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28644">ASTERISK-28644</a>: Stale comment in app_queue about ring_entry exception<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e1eb5e8dc269ca90c4503d8c8da058b3f3822eb9">[e1eb5e8dc2]</a> Walter Doekes -- app_queue: Fix old confusing comment about when the members are called</li>
</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28505">ASTERISK-28505</a>: app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream<br/>Reported by: Alexei Gradinari<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=052ab9d96604ba5f4e6666788bf93d357607fb95">[052ab9d966]</a> Alexei Gradinari -- app_voicemail/IMAP: check mailstream not NULL in leave_voicemail</li>
</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28637">ASTERISK-28637</a>: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e73893e53dddfc43f47dc6a451bc9e2a5387345">[3e73893e53]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28618">ASTERISK-28618</a>: bridge_softmix: hold not cleared when joining a softmix bridge<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b4610acfe476f111a52234e1fd2cab03b7d58f9">[8b4610acfe]</a> Kevin Harwell -- bridge_softmix: clear hold when joining a softmix bridge</li>
</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28566">ASTERISK-28566</a>: CDR backend unload problem during active call(s)<br/>Reported by: Marian Piater<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1dc3451a3464bdc52d03344535dd7ac7751777a2">[1dc3451a34]</a> Sean Bright -- cdr_mysql: Don't clean up on unload unless we can unregister from CDRs</li>
</ul><br><h4>Category: CDR/cdr_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28571">ASTERISK-28571</a>: cdr_pgsql: accesses obsolete (and finally removed) column<br/>Reported by: Christoph Moench-Tegeder<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79cc8ae3b809629df8f4739dfd537aa98d0c43ec">[79cc8ae3b8]</a> Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28615">ASTERISK-28615</a>: chan_dahdi: PRI span status may stay "Down, Active" after a short alarm<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3dd4c5459846b0bff4864f8089418a475c603db">[d3dd4c5459]</a> Frederic LE FOLL -- chan_dahdi: PRI span status may stay "Down, Active" after a short alarm</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28536">ASTERISK-28536</a>: Asterisk release candidates fail to build on FreeBSD<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5ff2f7a0169d30952f1ba9c5ff48027e6c950f03">[5ff2f7a016]</a> Guido Falsi -- chan_dahdi: Fix build with clang/llvm</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28525">ASTERISK-28525</a>: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e67c925027ec59154406f82cea4cb3934bdf2a7">[9e67c92502]</a> Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28578">ASTERISK-28578</a>: race condition on pjsip channelstats command<br/>Reported by: Salah Ahmed<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40acd7d1987e38d010430e013dd3dc0d54f8d4b9">[40acd7d198]</a> Salah Ahmed -- Crash during "pjsip show channelstats" execution</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28561">ASTERISK-28561</a>: Asterisk Deadlocks<br/>Reported by: Aheliotech<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae761c74738c04c11b72b539cc7e852bef6da260">[ae761c7473]</a> Joshua Colp -- pbx: deadlock when outgoing dialed channel hangs up too quickly</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28086">ASTERISK-28086</a>: chan_pjsip: Crash when initiating PlayDTMF over AMI<br/>Reported by: Jeremiah Gadd<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71f86e78b6e5694ef90baab9683f4f519d9540c1">[71f86e78b6]</a> lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28538">ASTERISK-28538</a>: chan_pjsip: Deadlock on fax detection<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d1baa3ae86b7adb6988ea73aa93f9966b714a4a">[4d1baa3ae8]</a> Joshua Colp -- chan_pjsip: Relock correct channel during "fax" redirect.</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28637">ASTERISK-28637</a>: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.<br/>Reported by: Frederic LE FOLL<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e73893e53dddfc43f47dc6a451bc9e2a5387345">[3e73893e53]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25592">ASTERISK-25592</a>: chan_unistim: Clang Warning: variable sized type not at end of a struct<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=056ddf76cee4f11dbb6529f9581c93301ebc3507">[056ddf76ce]</a> Igor Goncharovsky -- chan_unistim: Fix clang warning: variable sized type not at end of a struct</li>
</ul><br><h4>Category: Codecs/codec_resample</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28511">ASTERISK-28511</a>: codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32<br/>Reported by: Ruddy G<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75d6418d8eada0e59c55f7fcc5b680b31caf4303">[75d6418d8e]</a> Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34ab9964f57721fc4d9d54bc8cd7bab96bd52602">[34ab9964f5]</a> Sean Bright -- codec_resample: Upgrade speex_resample to fix up-sampling bug</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28487">ASTERISK-28487</a>: compile menuselect on gentoo<br/>Reported by: Kilburn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5f05eed70694e16f332e5433756ec27abd34e5c">[a5f05eed70]</a> Sean Bright -- menuselect: Fix curses build on Gentoo Linux</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28499">ASTERISK-28499</a>: translate: Crash when frame does not have a "src" field set<br/>Reported by: Gregory Massel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61c01df560f4cd193dd9bf2b0aaf7cc30b873a6a">[61c01df560]</a> Joshua Colp -- AST-2019-005 - translate: Don't assume all frames will have a src.</li>
</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23756">ASTERISK-23756</a>: setvar directive when used in template and a child of said template, results in duplicate variable names<br/>Reported by: Michael Goryainov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2fa296e7d41526f000841c083fcbd4d38a7fd776">[2fa296e7d4]</a> Michael Goryainov -- channels: Allow updating variable value</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28498">ASTERISK-28498</a>: cel / cdr: Event times may be incorrect<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=108b1abbd9d2ee0148b4833f581d9612ab46256c">[108b1abbd9]</a> Joshua Colp -- cdr / cel: Use event time at event creation instead of processing.</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28480">ASTERISK-28480</a>: json integer overflow in ssrc and timestamp<br/>Reported by: Salah Ahmed<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a305f2fdcb6c665c7bdfe779a370f09e1be35794">[a305f2fdcb]</a> Kevin Harwell -- various modules: json integer overflow</li>
</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28553">ASTERISK-28553</a>: stasis.c: Crash during unload<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=57fa6045711d2533de5ea08ec4d30b4ccfc885c2">[57fa604571]</a> Joshua Colp -- stasis: Pass bumped topic_all reference to proxy_dtor.</li>
</ul><br><h4>Category: Core/UDPTL</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28483">ASTERISK-28483</a>: packet lost on UDPTL wrap around<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44af3e9018108b01f219fc40a7fdb400a0d477d1">[44af3e9018]</a> Torrey Searle -- main/udptl.c: correctly handle udptl sequence wrap around</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26481">ASTERISK-26481</a>: FILE function grabs garbage along with read data when target line has no newline<br/>Reported by: Jonathan Harris<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92bb381d5d1878b9268e5da23a2e85cba3cdb715">[92bb381d5d]</a> Sean Bright -- func_env: Prevent FILE() from reading garbage at end-of-file</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28590">ASTERISK-28590</a>: utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument"<br/>Reported by: Speed Dial Dave<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3c56c7fa516944852ad2792b00410fc10448beb">[b3c56c7fa5]</a> Sean Bright -- utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28523">ASTERISK-28523</a>: Asterisk 16.5.0 Memory leak<br/>Reported by: Cyril Ramière<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f821e8107180e7c49c00c16ae8f998e12b51e79a">[f821e81071]</a> Kevin Harwell -- res_sorcery_memory_cache: stale item update leak</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28472">ASTERISK-28472</a>: Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV<br/>Reported by: Jonas Swiatek<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5daa9bbaee476fcf11223d174d90a987d96fb1a5">[5daa9bbaee]</a> Kevin Harwell -- srtp: Fix possible race condition, and add NULL checks</li>
</ul><br><h4>Category: PBX/pbx_config</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28534">ASTERISK-28534</a>: Segmentation fault when there is no priority for an extension<br/>Reported by: Timothy Vanderaerden<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d0edf2b3714bd1096902d15d240c2a43c270bd5">[8d0edf2b37]</a> Sean Bright -- pbx: Prevent Realtime switch crash on invalid priority</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28585">ASTERISK-28585</a>: ari/resource_events: Crash in event session cleanup<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e37d5461095e93ad274dd9893d21750b8d2cacc1">[e37d546109]</a> Joshua Colp -- res_ari_events: Add module reference when a WebSocket is open.</li>
</ul><br><h4>Category: Resources/res_calendar_exchange</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28572">ASTERISK-28572</a>: Memory leaks in res_calendar_exchange and res_calendar_icalendar<br/>Reported by: Yoooooo Ha<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3792e1288d481718875fde9da35b9332301d766">[b3792e1288]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
</ul><br><h4>Category: Resources/res_calendar_icalendar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28572">ASTERISK-28572</a>: Memory leaks in res_calendar_exchange and res_calendar_icalendar<br/>Reported by: Yoooooo Ha<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3792e1288d481718875fde9da35b9332301d766">[b3792e1288]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28631">ASTERISK-28631</a>: res_parking: Doesn't park when parkee and parker are the same<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41d58a4ce2ea34345c6c9645238eb80ae39e15ac">[41d58a4ce2]</a> Joshua Colp -- parking: Fall back to parker channel name even if it matches parkee.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28616">ASTERISK-28616</a>: parking: Deadlock when multi call parking<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de433cdcaf64070164300a51ab8ab0d0c596c71e">[de433cdcaf]</a> Joshua Colp -- parking: Fix case where we can't get the parker.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d638d9c6c6b9ffe8feb0ee35466a6d884a5d04e4">[d638d9c6c6]</a> Joshua Colp -- parking: Use channel snapshot instead of channel.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28641">ASTERISK-28641</a>: res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e057eb9d2c3572373c95f5b272ac8950efcbfec">[4e057eb9d2]</a> Sean Bright -- res_pjsip_registrar.c: Prevent potential double free if AOR is not found</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28544">ASTERISK-28544</a>: Wrong contact representation in ipv6 mode<br/>Reported by: Jørgen H<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6527eb8213e185e3b5e3a194764d688f4d73a088">[6527eb8213]</a> Sean Bright -- res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28521">ASTERISK-28521</a>: pjsip: Memory Leak<br/>Reported by: Mark<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c0435f854d34f2723fa744ee599845b25bcc5f8">[7c0435f854]</a> George Joseph -- pjproject_bundled: Revert pjproject 2.9 commits causing leaks</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28228">ASTERISK-28228</a>: res_pjsip: pjsip show contacts prints double entries<br/>Reported by: Ian Jones<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20459d4cacc94335b7e39ed696f336c521f872ba">[20459d4cac]</a> Joshua Colp -- res_pjsip: Fix multiple of the same contact in "pjsip show contacts".</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28575">ASTERISK-28575</a>: MWI Send Notify Crash on 16.6<br/>Reported by: Joshua Elson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45c0d991857ec1ee61d12bc26a106cc8d5cc84a3">[45c0d99185]</a> Kevin Harwell -- res_pjsip_mwi: potential double unref, and potential unwanted double link</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28552">ASTERISK-28552</a>: res_pjsip_mwi: Frack during unload on unsolicited_mwi container<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=996fc40e2b489c7df13567d2255d8238c0cad1bd">[996fc40e2b]</a> Kevin Harwell -- res_pjsip_mwi: use an ao2_global object for mwi containers</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28624">ASTERISK-28624</a>: res_pjsip_outbound_registration: add SRV failover<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c99930375bb73072a20e30893fecd1c382f5671">[8c99930375]</a> Kevin Harwell -- res_pjsip_outbound_registration: add support for SRV failover</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28521">ASTERISK-28521</a>: pjsip: Memory Leak<br/>Reported by: Mark<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c0435f854d34f2723fa744ee599845b25bcc5f8">[7c0435f854]</a> George Joseph -- pjproject_bundled: Revert pjproject 2.9 commits causing leaks</li>
</ul><br><h4>Category: Resources/res_pjsip_path</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28463">ASTERISK-28463</a>: res_pjsip_path: Crash when invalid contact is configured<br/>Reported by: Juan Martin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51cf060c6ccca4d545ab4723f412318388b185f1">[51cf060c6c]</a> Sean Bright -- res_pjsip_registrar: Validate Contact URI before adding to responses</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28445">ASTERISK-28445</a>: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled<br/>Reported by: Bernhard Schmidt<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d56adf8fbbbaa6c05c547eb482f1d154ec006d4">[4d56adf8fb]</a> Sean Bright -- res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28086">ASTERISK-28086</a>: chan_pjsip: Crash when initiating PlayDTMF over AMI<br/>Reported by: Jeremiah Gadd<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71f86e78b6e5694ef90baab9683f4f519d9540c1">[71f86e78b6]</a> lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel</li>
</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28621">ASTERISK-28621</a>: Enforce T.38 error correction mode at 200 ok received <br/>Reported by: Salah Ahmed<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ac0299bfbdfc851040ab2e8136e0078378ea19a">[4ac0299bfb]</a> Salah Ahmed -- res_pjsip_t38: T.38 error correction mode selection at 200 ok received</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28576">ASTERISK-28576</a>: res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match<br/>Reported by: Joshua Elson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c486e7edf9afae2e649aba47be2464e9a25db38">[0c486e7edf]</a> Joshua Colp -- res_rtp_asterisk: Always return provided DTLS packet length.</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7202624b3b46bf013a08f26480a60fa2c515a80f">[7202624b3b]</a> George Joseph -- stasis: Don't hold app_registry and session locks unnecessarily</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28574">ASTERISK-28574</a>: pjproject fails to build on 16.6.0, works on 16.5<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2652bda3a0bb0dd2b444a4e224ff3dcea2eb6e93">[2652bda3a0]</a> George Joseph -- pjproject_bundled: Replace earlier reverts with official fixes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28509">ASTERISK-28509</a>: PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters<br/>Reported by: Dan Cropp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1d38e19a267e06042c5c98d6204bd14a0a0c4e4">[a1d38e19a2]</a> Dan Cropp -- pjproject: Configurable setting for cnonce to include hyphens or not</li>
</ul><br><h3>New Feature</h3><h4>Category: Applications/app_senddtmf</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28614">ASTERISK-28614</a>: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending"<br/>Reported by: lvl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6345a002284e68ae5637088678a4491963d28845">[6345a00228]</a> lvl -- app_senddtmf: Add receive mode to AMI Action PlayDTMF</li>
</ul><br><h4>Category: Core/Jitterbuffer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28533">ASTERISK-28533</a>: func_jitterbuffer: Add support for video synchronization<br/>Reported by: Joshua C. Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=926053d7bdf6245d9081550eadd5c61ed2aeb784">[926053d7bd]</a> Joshua Colp -- func_jitterbuffer: Add audio/video sync support.</li>
</ul><br><h4>Category: Functions/func_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28613">ASTERISK-28613</a>: func_curl: CURLOPT cannot set Content-Type header<br/>Reported by: Martin Tomec<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d579ec9cdf85ab31c1b7d9c8391d41b4ec513fc9">[d579ec9cdf]</a> Martin Tomec -- func_curl.c: Support custom http headers</li>
</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17808">ASTERISK-17808</a>: [patch] Unregister a realtime moh class<br/>Reported by: Byron Clark<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e26136ee6a2d09dfe26528d109eedd12a7948ef">[9e26136ee6]</a> sungtae kim -- res_musiconhold: Added unregister realtime moh class</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28489">ASTERISK-28489</a>: Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain<br/>Reported by: Stas Kobzar<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a246c2a69cd7f160d80c31058e99e9f99de6746">[3a246c2a69]</a> Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN</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: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28626">ASTERISK-28626</a>: Missing arguments in PJSIP_CONTACT function documentation<br/>Reported by: Pascal Cadotte Michaud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=450173a0ae3aace0bbdcbfbcd336c5069747b0d1">[450173a0ae]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</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=08a6e8c55315deedeae5e7c1d0528286f1c8a153">08a6e8c553</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.1.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a89784b7820c301e8b438d140c836db84d3f2b9">6a89784b78</a></td><td>Joshua Colp</td><td>Revert "PJSIP_CONTACT: add missing argument documentation"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f26e5bacc008fc5335cb260f75ae73ee3fcbb67a">f26e5bacc0</a></td><td>Sean Bright</td><td>res_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88150323a235efeb051dc0cae9ef615dd34e86d1">88150323a2</a></td><td>Thomas Arimont</td><td>channel.c: Resolve issue with receiving SIP INFO packets for DTMF</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b15a1c6393b380ad86daa40a08491d880f94e68">5b15a1c639</a></td><td>George Joseph</td><td>CI: Turn off shallow cloning altogether</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc59e21409b22c109560b92a6073bffc30006f7c">cc59e21409</a></td><td>Sean Bright</td><td>media_cache.c: Various CLI improvements</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a92e6b576334ac926d7d498cfab6081d16e702e">2a92e6b576</a></td><td>George Joseph</td><td>CI: Fix missing script block in jenkinsfiles</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0d1ce50afd25a1269e680b90c8bb612bd543565">f0d1ce50af</a></td><td>George Joseph</td><td>CI: Fix missing script block in jenkinsfiles</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46dceab33f5b54ee0f222646bb92654c07efe01f">46dceab33f</a></td><td>George Joseph</td><td>CI: Increase clone depth and do better cleanup</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=76ef36fafcb9a0250e036eb8172681b506a81ee0">76ef36fafc</a></td><td>Sean Bright</td><td>res_pjsip_registrar: Fix uninitlized variable warning</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=649733612d84fbc8566822513dfe9b6fc88c778c">649733612d</a></td><td>Alexei Gradinari</td><td>serializer: set high/low alert levels on whole pool</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bc6fa0fbd82f4a57e492f0f92b2acd6454a47b7">8bc6fa0fbd</a></td><td>Kevin Harwell</td><td>various files - fix some alerts raised by lgtm code analysis</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea3daa94c8c04e2037c5fd801bcd703a8209ac94">ea3daa94c8</a></td><td>Kevin Harwell</td><td>res_pjsip_session: initialize pending's topology to endpoint's</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d665091a36882070dae8ac733aa551f260488ad">2d665091a3</a></td><td>George Joseph</td><td>ExternalMedia: Change return object from ExternalMedia to Channel</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b8ae799ca9da487e87406edc932a575a600ac735">b8ae799ca9</a></td><td>Joshua Colp</td><td>res_rtp_asterisk: Remove a log message that slipped in.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ba688e6891e069761ee996baab507e00e4da8278">ba688e6891</a></td><td>Joshua Colp</td><td>test_res_rtp: Enable FIR and REMB nominal tests.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c84135d2a3527b383280a6732dcee5d30a5a5486">c84135d2a3</a></td><td>Chris Savinovich</td><td>test_taskprocessor.c: Fix test failure on Ubuntu</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37ec88c4c8bd71d9a50209966274bbf8559562fa">37ec88c4c8</a></td><td>Kevin Harwell</td><td>serializer: move/add asterisk serializer pool functionality</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=299ba78b0955f17bfac5703b891d0d19466f99c1">299ba78b09</a></td><td>Kevin Harwell</td><td>res_pjsip/res_pjsip_mwi: use centralized serializer pools</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25fbe7979396d83da3bef91353894e0862bb7897">25fbe79793</a></td><td>Corey Farrell</td><td>stasis_state: Create internal stasis_state_proxy object.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b47d4774dd75fda72cb2a8ebc39624b2e575a3c">4b47d4774d</a></td><td>Alexei Gradinari</td><td>res_pjsip_pubsub: add endpoint to some warning</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d223419bcddaf0a695ee90fdb70d4f8fd721090e">d223419bcd</a></td><td>Jonathan Rose</td><td>basic-pbx: Bring forward queue configuration from 13</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8269fcbf0343b7515df2073cd8b847ce4b24aa6b">8269fcbf03</a></td><td>Ben Ford</td><td>taskprocessor.c: Added "like" support to 'core show taskprocessors'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37139e16a554e99b5cc8457681cf04e20c585fb4">37139e16a5</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.0.0-rc2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7550a82fe0cfbcf1309bb410293c8086a13a8bd2">7550a82fe0</a></td><td>Sean Bright</td><td>res_musiconhold: Add new 'playlist' mode</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=175a7ccac76529b8f238f2ad61b8cd4cba0c0d10">175a7ccac7</a></td><td>Kevin Harwell</td><td>res_pjsip_pubsub: change warning to debug</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5bedd4a9b45bb2d7ded09ad6d0b3561e2d091711">5bedd4a9b4</a></td><td>Corey Farrell</td><td>core: Fix ABI mismatch of ao2_global_obj.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f181dfc0657cfe8e2558eed68665a9ce8f0d4a54">f181dfc065</a></td><td>Ben Ford</td><td>taskprocessor.c: Add CLI commands to reset taskprocessor stats.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec712cfab24552406a7e104bd8ab02201f78c233">ec712cfab2</a></td><td>Corey Farrell</td><td>core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1c5753b78344d2fc6b4d209116dea1f50d76bcd">b1c5753b78</a></td><td>George Joseph</td><td>astmm.c: Display backtrace with memory show allocations</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af0ccbc45e62efc92bccabcd7e19593ae7b4955b">af0ccbc45e</a></td><td>Corey Farrell</td><td>stasis: refcounter.py can incorrectly report skewed objects.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd7fc3d7a5fcabb9dbf925fdcdf51b4fe50ece82">fd7fc3d7a5</a></td><td>Corey Farrell</td><td>stasis: Fix leaks</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78f329dad9a84312909cf517945ae1fe71b801d9">78f329dad9</a></td><td>Corey Farrell</td><td>app_voicemail: Fix module unload leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d960195c2b28d6c78794f699b07dc6f54460d2f">1d960195c2</a></td><td>Ben Ford</td><td>res_rtp_asterisk.c: Send RTCP as compound packets.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5dad4f4ea1bce2e823ed72e075b51d637673670f">5dad4f4ea1</a></td><td>Ben Ford</td><td>res_rtp: Add unit tests for RTCP stats.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44068cc6bf802742ce8a0283351781bbcd9d7690">44068cc6bf</a></td><td>George Joseph</td><td>ARI: External Media</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c060ef7dee4bdd56504478f3ab74ae56646e2b6f">c060ef7dee</a></td><td>George Joseph</td><td>chan_sip: Update links referenced in deprecation notice</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d99710fa683fbceec1ff12f7d52ecd5549f08e0">7d99710fa6</a></td><td>Chris-Savinovich</td><td>test_utils.c: Skip test adsi_loaded_test if module not loaded.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=95b3c9839da7a4cc01feeeab2b65350bc8e0ce40">95b3c9839d</a></td><td>Igor Goncharovsky</td><td>chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7f1f7eefd13c535ed90b30fa6122dbe43f076a4">e7f1f7eefd</a></td><td>Igor Goncharovsky</td><td>chan_unistim: Fix RTP port byte order for big-endian arch</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0b0c7540c2d9623ccbe73390f95fd3f2573eb3e">a0b0c7540c</a></td><td>Alexei Gradinari</td><td>Fix misname 'res_external_mwi' to 'res_mwi_external' in comments.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=581ffdf6a98c53ec41d640161420bdb07f4d3d9b">581ffdf6a9</a></td><td>George Joseph</td><td>chan_rtp: Accept hostname as well as ip address as destination</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3d9f6ef6cf7541676631909f3852c6b1b5030fb">b3d9f6ef6c</a></td><td>George Joseph</td><td>dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6ab42df2641ad8e32849d7bb608bd35f4fa313e">f6ab42df26</a></td><td>George Joseph</td><td>res_ari.c: Prefer exact handler match over wildcard</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e123d1ccc581d319d5efbe647920b738f1af6505">e123d1ccc5</a></td><td>Sean Bright</td><td>audiohook.c: Substitute silence for unavailable audio frames</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5489ad5574331c909075539dc9ccaef4acebb477">5489ad5574</a></td><td>George Joseph</td><td>CI: Escape backslashes in printenv/sort/tr</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa44e723c56f65b363955f5c945b9be97ed545cc">aa44e723c5</a></td><td>George Joseph</td><td>CI: Add "throttle" label and "skip_gate" capability</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=865214079447f987e3a3d1b1a31cea2524547614">8652140794</a></td><td>George Joseph</td><td>CI: Make node labels job-specific</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87f8ca2812b45e2c1eb8f5897b35575d2ded4e23">87f8ca2812</a></td><td>Sean Bright</td><td>app_voicemail: Remove extra menuselect build options</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1976db6ffd912dfb9ac0bc42748ae9c687171c7a">1976db6ffd</a></td><td>Sean Bright</td><td>res_musiconhold: Use a vector instead of custom array allocation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c748187bbfc693cce2ffbdb80cdf8f0310d0ad76">c748187bbf</a></td><td>Sean Bright</td><td>manager: Send fewer packets</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1e0c9d1b819ccda353412e6cf8e29ff273f0145">f1e0c9d1b8</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 17.0.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f2ad5a55d5d44550e21a817c35c46a6baa7f916">4f2ad5a55d</a></td><td>George Joseph</td><td>doc: Add "master-only" flag back to the CHANGES and UPGRADE files</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a66ccb90ac7175b33dea0590aa9482e8afb6078b">a66ccb90ac</a></td><td>Sean Bright</td><td>res_musiconhold: Use ast_pipe_nonblock() wrapper</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e821a637a84c7ee2e6549c44c2f1ca6637daae18">e821a637a8</a></td><td>George Joseph</td><td>loader.c: Fix possible SEGV when a module fails to register</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 |86152 ----------
asterisk-17.0.0-summary.html | 1265
asterisk-17.0.0-summary.txt | 2973
b/CHANGES | 94
b/README-SERIOUSLY.bestpractices.md | 4
b/UPGRADE.txt | 10
b/addons/cdr_mysql.c | 18
b/addons/chan_ooh323.c | 7
b/apps/app_amd.c | 24
b/apps/app_cdr.c | 8
b/apps/app_chanisavail.c | 3
b/apps/app_dictate.c | 4
b/apps/app_followme.c | 10
b/apps/app_meetme.c | 2
b/apps/app_minivm.c | 3
b/apps/app_mixmonitor.c | 13
b/apps/app_playback.c | 9
b/apps/app_queue.c | 6
b/apps/app_readexten.c | 3
b/apps/app_senddtmf.c | 13
b/apps/app_voicemail.c | 217
b/bridges/bridge_native_rtp.c | 5
b/bridges/bridge_softmix.c | 7
b/cdr/cdr_pgsql.c | 2
b/cel/cel_pgsql.c | 2
b/channels/chan_dahdi.c | 35
b/channels/chan_dahdi.h | 18
b/channels/chan_iax2.c | 16
b/channels/chan_motif.c | 9
b/channels/chan_pjsip.c | 24
b/channels/chan_rtp.c | 19
b/channels/chan_sip.c | 39
b/channels/chan_unistim.c | 174
b/channels/pjsip/cli_commands.c | 13
b/channels/pjsip/dialplan_functions.c | 65
b/channels/pjsip/include/dialplan_functions.h | 25
b/channels/sig_pri.c | 19
b/codecs/Makefile | 3
b/codecs/ex_alaw.h | 5
b/codecs/ex_g722.h | 5
b/codecs/ex_ulaw.h | 5
b/codecs/speex/arch.h | 13
b/codecs/speex/fixed_generic.h | 4
b/codecs/speex/resample.c | 332
b/codecs/speex/speex_resampler.h | 4
b/configs/basic-pbx/extensions.conf | 14
b/configs/basic-pbx/modules.conf | 1
b/configs/basic-pbx/queues.conf | 19
b/configs/samples/extconfig.conf.sample | 1
b/configs/samples/musiconhold.conf.sample | 23
b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py | 54
b/formats/format_g726.c | 16
b/formats/msgsm.h | 4
b/funcs/func_curl.c | 37
b/funcs/func_env.c | 5
b/funcs/func_jitterbuffer.c | 19
b/include/asterisk/abstract_jb.h | 4
b/include/asterisk/ari.h | 2
b/include/asterisk/astobj2.h | 5
b/include/asterisk/audiohook.h | 2
b/include/asterisk/calendar.h | 4
b/include/asterisk/channel_internal.h | 5
b/include/asterisk/config.h | 18
b/include/asterisk/config_options.h | 2
b/include/asterisk/dns_core.h | 22
b/include/asterisk/dns_internal.h | 5
b/include/asterisk/format_cache.h | 5
b/include/asterisk/max_forwards.h | 1
b/include/asterisk/mixmonitor.h | 5
b/include/asterisk/parking.h | 5
b/include/asterisk/res_pjsip.h | 5
b/include/asterisk/res_pjsip_presence_xml.h | 5
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/rtp_engine.h | 111
b/include/asterisk/serializer.h | 85
b/include/asterisk/slin.h | 5
b/include/asterisk/taskprocessor.h | 9
b/include/asterisk/utils.h | 9
b/main/abstract_jb.c | 178
b/main/app.c | 9
b/main/asterisk.c | 15
b/main/astmm.c | 23
b/main/astobj2.c | 88
b/main/astobj2_container.c | 24
b/main/astobj2_global.c | 97
b/main/astobj2_hash.c | 21
b/main/astobj2_rbtree.c | 13
b/main/audiohook.c | 11
b/main/channel.c | 105
b/main/codec_builtin.c | 8
b/main/config.c | 16
b/main/dns_core.c | 72
b/main/event.c | 17
b/main/file.c | 37
b/main/format_cache.c | 8
b/main/indications.c | 6
b/main/manager.c | 92
b/main/media_cache.c | 47
b/main/pbx.c | 5
b/main/pbx_variables.c | 23
b/main/rtp_engine.c | 137
b/main/serializer.c | 189
b/main/stasis.c | 22
b/main/stasis_cache.c | 10
b/main/stasis_state.c | 298
b/main/taskprocessor.c | 219
b/res/ari/config.c | 10
b/res/ari/resource_channels.c | 158
b/res/ari/resource_channels.h | 42
b/res/ari/resource_events.c | 10
b/res/parking/parking_bridge.c | 36
b/res/parking/parking_bridge_features.c | 2
b/res/parking/res_parking.h | 5
b/res/res_ari_channels.c | 135
b/res/res_ari_events.c | 2
b/res/res_calendar_ews.c | 1
b/res/res_calendar_exchange.c | 1
b/res/res_calendar_icalendar.c | 1
b/res/res_config_curl.c | 5
b/res/res_config_pgsql.c | 2
b/res/res_musiconhold.c | 294
b/res/res_phoneprov.c | 6
b/res/res_pjsip.c | 83
b/res/res_pjsip/config_system.c | 2
b/res/res_pjsip/config_transport.c | 17
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/pjsip_resolver.c | 4
b/res/res_pjsip_endpoint_identifier_ip.c | 18
b/res/res_pjsip_mwi.c | 339
b/res/res_pjsip_outbound_registration.c | 17
b/res/res_pjsip_pubsub.c | 20
b/res/res_pjsip_registrar.c | 55
b/res/res_pjsip_session.c | 19
b/res/res_pjsip_t38.c | 40
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_resolver_unbound.c | 6
b/res/res_rtp_asterisk.c | 276
b/res/res_stasis.c | 43
b/res/stasis/command.c | 2
b/res/stasis/control.c | 2
b/rest-api-templates/res_ari_resource.c.mustache | 2
b/rest-api/api-docs/channels.json | 125
b/tests/CI/gates.jenkinsfile | 12
b/tests/CI/periodics-daily.jenkinsfile | 11
b/tests/CI/ref_debug.jenkinsfile | 9
b/tests/CI/unittests.jenkinsfile | 9
b/tests/test_data_buffer.c | 2
b/tests/test_res_rtp.c | 516
b/tests/test_taskprocessor.c | 28
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1255
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1361
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
157 files changed, 4528 insertions(+), 94603 deletions(-)</pre><br></html>

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asterisk-17.1.0-rc1-summary.txt
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@@ -1,945 +0,0 @@
Release Summary
asterisk-17.1.0-rc1
Date: 2019-12-12
<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-17.0.0.
----------------------------------------------------------------------
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
23 Sean Bright 1 tests/test_utils.c. 5 Kevin Harwell
21 George Joseph 5 Joshua C. Colp
16 Joshua Colp 4 Frederic LE FOLL
13 Kevin Harwell 3 Salah Ahmed
6 Corey Farrell 3 Ross Beer
5 Alexei Gradinari 2 Alexei Gradinari
5 Ben Ford 2 Torrey Searle
4 Frederic LE FOLL 2 Ross Beer
3 Asterisk Development 2 Joshua Elson
Team 2 Ruddy G
3 Igor Goncharovsky 1 Michael Cargile
2 Salah Ahmed 1 Walter Doekes
2 Torrey Searle 1 Martin Tomec
2 lvl 1 Chris Savinovich
1 Thomas Arimont (license 1 Byron Clark
5525) 1 Niklas Larsson
1 Pascal Cadotte Michaud 1 Jonas Swiatek
1 Martin Tomec 1 Yoooooo Ha
1 Walter Doekes 1 Michael
1 Stas Kobzar 1 Eliel Sardañons
1 Jonathan Rose 1 Guido Falsi
1 Michael Goryainov 1 Gregory Massel
1 Chris-Savinovich 1 Dan Cropp
1 Michael Cargile 1 Jeremiah Gadd
1 Chris Savinovich 1 Bernhard Schmidt
1 sungtae kim 1 Stas Kobzar
1 Florian Floimair 1 Marian Piater
1 cmaj 1 Pascal Cadotte Michaud
1 Christoph Moench-Tegeder 1 Kilburn
1 Dan Cropp 1 Michael Goryainov
1 Guido Falsi 1 Bernhard Schmidt
1 Ian Jones
1 Alexander Traud
1 Aheliotech
1 Dan Cropp
1 Mark
1 Timothy Vanderaerden
1 Niklas Larsson
1 Andrey V. T.
1 Christoph
Moench-Tegeder
1 Florian Floimair
1 Speed Dial Dave
1 Jonathan Harris
1 Daniel
1 Sam Banks
1 George Joseph
1 Eliel Sardañons
1 Alexander Traud
1 Cyril Ramière
1 Jørgen H
1 cmaj
1 Juan Martin
1 lvl
----------------------------------------------------------------------
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.
Security
Category: Channels/chan_sip/General
ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter
Addr of a peer
Reported by: Andrey V. T.
* [665a94cb76] Ben Ford -- chan_sip.c: Prevent address change on
unauthenticated SIP request.
Category: Core/ManagerInterface
ASTERISK-28580: Bypass SYSTEM write permission in manager action allows
system commands execution
Reported by: Eliel Sardañons
* [6b1ba58967] George Joseph -- manager.c: Prevent the Originate action
from running the Originate app
Category: Resources/res_pjsip_t38
ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream
causes crash
Reported by: Alexei Gradinari
* [9d4f1e8ebe] Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add
NULL checks before using session media
Improvement
Category: Applications/app_voicemail
ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any
.lock files in the voice mail directory on startup.
Reported by: Michael
* [b903994987] Sean Bright -- Revert "app_voicemail: Cleanup stale lock
files on module load"
Category: Applications/app_voicemail/ODBC
ASTERISK-22192: [patch] Allow voicemail forwards with ODBC backend when
format differs from attachfmt column
Reported by: cmaj
* [aa0973f868] cmaj -- app_voicemail.c: Support multiple file formats
for forwarded messages.
Category: Core/CodecInterface
ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
Reported by: Florian Floimair
* [d7a3e4f5cf] Florian Floimair -- core: Add H.265/HEVC passthrough
support
Category: Documentation
ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
Reported by: Sam Banks
* [4bc1c170cd] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling
correetions.
Category: Resources/res_pjsip
ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold
re-invites
Reported by: Torrey Searle
* [55b760d762] Torrey Searle -- channel/chan_pjsip: add dialplan
function for music on hold
Category: Resources/res_pjsip_outbound_registration
ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
Reported by: Daniel
* [eea2d499f4] Joshua Colp -- res_pjsip_outbound_registration: Extend
documentation for "max_retries".
Bug
Category: .Release/Targets
ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register
Reported by: Chris Savinovich
* [a36fb473fe] Kevin Harwell -- res_pjsip_mwi: add better handling of
solicited vs unsolicited subscriptions
Category: Applications/app_amd
ASTERISK-28608: app_amd: Use time calculation to calculate timeout
Reported by: Michael Cargile
* [e23b2856d0] Michael Cargile -- app_amd: Fixed timeout issue
Category: Applications/app_chanisavail
ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in
cdr.conf
Reported by: Frederic LE FOLL
* [50997de887] Frederic LE FOLL -- ChanIsAvail() generates a CDR when
unanswered=yes in cdr.conf.
Category: Applications/app_meetme
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
17.0.0
Reported by: George Joseph
* [36b28c98dd] George Joseph -- Build: Fix compile issues with seldom
used modules
Category: Applications/app_queue
ASTERISK-28644: Stale comment in app_queue about ring_entry exception
Reported by: Walter Doekes
* [e1eb5e8dc2] Walter Doekes -- app_queue: Fix old confusing comment
about when the members are called
Category: Applications/app_voicemail/IMAP
ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because
not checking mailstream
Reported by: Alexei Gradinari
* [052ab9d966] Alexei Gradinari -- app_voicemail/IMAP: check mailstream
not NULL in leave_voicemail
Category: Bridges/bridge_native_rtp
ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
check failure when negociated ptime is not default ptime.
Reported by: Frederic LE FOLL
* [3e73893e53] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
directmedia for ptime other than default ptime.
Category: Bridges/bridge_softmix
ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix
bridge
Reported by: Kevin Harwell
* [8b4610acfe] Kevin Harwell -- bridge_softmix: clear hold when joining
a softmix bridge
Category: CDR/General
ASTERISK-28566: CDR backend unload problem during active call(s)
Reported by: Marian Piater
* [1dc3451a34] Sean Bright -- cdr_mysql: Don't clean up on unload unless
we can unregister from CDRs
Category: CDR/cdr_pgsql
ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column
Reported by: Christoph Moench-Tegeder
* [79cc8ae3b8] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql
res_config_pgsql: compatibility with PostgreSQL 12
Category: Channels/chan_dahdi
ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after
a short alarm
Reported by: Frederic LE FOLL
* [d3dd4c5459] Frederic LE FOLL -- chan_dahdi: PRI span status may stay
"Down, Active" after a short alarm
ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD
Reported by: Guido Falsi
* [5ff2f7a016] Guido Falsi -- chan_dahdi: Fix build with clang/llvm
ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
hangs up
Reported by: Frederic LE FOLL
* [9e67c92502] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource)
when a PRI channel hangs up
Category: Channels/chan_pjsip
ASTERISK-28578: race condition on pjsip channelstats command
Reported by: Salah Ahmed
* [40acd7d198] Salah Ahmed -- Crash during "pjsip show channelstats"
execution
ASTERISK-28561: Asterisk Deadlocks
Reported by: Aheliotech
* [ae761c7473] Joshua Colp -- pbx: deadlock when outgoing dialed channel
hangs up too quickly
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
Reported by: Jeremiah Gadd
* [71f86e78b6] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
on hungup channel
ASTERISK-28538: chan_pjsip: Deadlock on fax detection
Reported by: Joshua C. Colp
* [4d1baa3ae8] Joshua Colp -- chan_pjsip: Relock correct channel during
"fax" redirect.
Category: Channels/chan_sip/General
ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
check failure when negociated ptime is not default ptime.
Reported by: Frederic LE FOLL
* [3e73893e53] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
directmedia for ptime other than default ptime.
Category: Channels/chan_unistim
ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at
end of a struct
Reported by: Alexander Traud
* [056ddf76ce] Igor Goncharovsky -- chan_unistim: Fix clang warning:
variable sized type not at end of a struct
Category: Codecs/codec_resample
ASTERISK-28511: codec_resample: Bad sound quality when up sampling from
SLIN16 to SLIN32
Reported by: Ruddy G
* [75d6418d8e] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is
defined when necessary
* [34ab9964f5] Sean Bright -- codec_resample: Upgrade speex_resample to
fix up-sampling bug
Category: Core/BuildSystem
ASTERISK-28487: compile menuselect on gentoo
Reported by: Kilburn
* [a5f05eed70] Sean Bright -- menuselect: Fix curses build on Gentoo
Linux
Category: Core/Channels
ASTERISK-28499: translate: Crash when frame does not have a "src" field
set
Reported by: Gregory Massel
* [61c01df560] Joshua Colp -- AST-2019-005 - translate: Don't assume all
frames will have a src.
Category: Core/Configuration
ASTERISK-23756: setvar directive when used in template and a child of said
template, results in duplicate variable names
Reported by: Michael Goryainov
* [2fa296e7d4] Michael Goryainov -- channels: Allow updating variable
value
Category: Core/General
ASTERISK-28498: cel / cdr: Event times may be incorrect
Reported by: Joshua C. Colp
* [108b1abbd9] Joshua Colp -- cdr / cel: Use event time at event
creation instead of processing.
Category: Core/RTP
ASTERISK-28480: json integer overflow in ssrc and timestamp
Reported by: Salah Ahmed
* [a305f2fdcb] Kevin Harwell -- various modules: json integer overflow
Category: Core/Stasis
ASTERISK-28553: stasis.c: Crash during unload
Reported by: Kevin Harwell
* [57fa604571] Joshua Colp -- stasis: Pass bumped topic_all reference to
proxy_dtor.
Category: Core/UDPTL
ASTERISK-28483: packet lost on UDPTL wrap around
Reported by: Torrey Searle
* [44af3e9018] Torrey Searle -- main/udptl.c: correctly handle udptl
sequence wrap around
Category: Functions/General
ASTERISK-26481: FILE function grabs garbage along with read data when
target line has no newline
Reported by: Jonathan Harris
* [92bb381d5d] Sean Bright -- func_env: Prevent FILE() from reading
garbage at end-of-file
Category: General
ASTERISK-28590: utils.c throws repeated warnings;
"pthread_attr_setstacksize: Invalid argument"
Reported by: Speed Dial Dave
* [b3c56c7fa5] Sean Bright -- utils.h: Set lower bound for thread stack
size to PTHREAD_STACK_MIN
ASTERISK-28523: Asterisk 16.5.0 Memory leak
Reported by: Cyril Ramière
* [f821e81071] Kevin Harwell -- res_sorcery_memory_cache: stale item
update leak
ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to
srtp_protect/unprotect causing SEGV
Reported by: Jonas Swiatek
* [5daa9bbaee] Kevin Harwell -- srtp: Fix possible race condition, and
add NULL checks
Category: PBX/pbx_config
ASTERISK-28534: Segmentation fault when there is no priority for an
extension
Reported by: Timothy Vanderaerden
* [8d0edf2b37] Sean Bright -- pbx: Prevent Realtime switch crash on
invalid priority
Category: Resources/res_ari
ASTERISK-28585: ari/resource_events: Crash in event session cleanup
Reported by: Kevin Harwell
* [e37d546109] Joshua Colp -- res_ari_events: Add module reference when
a WebSocket is open.
Category: Resources/res_calendar_exchange
ASTERISK-28572: Memory leaks in res_calendar_exchange and
res_calendar_icalendar
Reported by: Yoooooo Ha
* [b3792e1288] Sean Bright -- res_calendar: Resolve memory leak on
calendar destruction
Category: Resources/res_calendar_icalendar
ASTERISK-28572: Memory leaks in res_calendar_exchange and
res_calendar_icalendar
Reported by: Yoooooo Ha
* [b3792e1288] Sean Bright -- res_calendar: Resolve memory leak on
calendar destruction
Category: Resources/res_parking
ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the
same
Reported by: Ross Beer
* [41d58a4ce2] Joshua Colp -- parking: Fall back to parker channel name
even if it matches parkee.
ASTERISK-28616: parking: Deadlock when multi call parking
Reported by: Joshua C. Colp
* [de433cdcaf] Joshua Colp -- parking: Fix case where we can't get the
parker.
* [d638d9c6c6] Joshua Colp -- parking: Use channel snapshot instead of
channel.
Category: Resources/res_pjsip
ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR
points to a not existent AOR
Reported by: Ross Beer
* [4e057eb9d2] Sean Bright -- res_pjsip_registrar.c: Prevent potential
double free if AOR is not found
ASTERISK-28544: Wrong contact representation in ipv6 mode
Reported by: Jørgen H
* [6527eb8213] Sean Bright -- res_pjsip_transport_websocket: Don't put
brackets around local_name if IPv6
ASTERISK-28521: pjsip: Memory Leak
Reported by: Mark
* [7c0435f854] George Joseph -- pjproject_bundled: Revert pjproject 2.9
commits causing leaks
ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries
Reported by: Ian Jones
* [20459d4cac] Joshua Colp -- res_pjsip: Fix multiple of the same
contact in "pjsip show contacts".
Category: Resources/res_pjsip_mwi
ASTERISK-28575: MWI Send Notify Crash on 16.6
Reported by: Joshua Elson
* [45c0d99185] Kevin Harwell -- res_pjsip_mwi: potential double unref,
and potential unwanted double link
ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi
container
Reported by: Kevin Harwell
* [996fc40e2b] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object
for mwi containers
Category: Resources/res_pjsip_outbound_registration
ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
Reported by: Kevin Harwell
* [8c99930375] Kevin Harwell -- res_pjsip_outbound_registration: add
support for SRV failover
ASTERISK-28521: pjsip: Memory Leak
Reported by: Mark
* [7c0435f854] George Joseph -- pjproject_bundled: Revert pjproject 2.9
commits causing leaks
Category: Resources/res_pjsip_path
ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured
Reported by: Juan Martin
* [51cf060c6c] Sean Bright -- res_pjsip_registrar: Validate Contact URI
before adding to responses
Category: Resources/res_pjsip_session
ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
hangup when TEST_FRAMEWORK enabled
Reported by: Bernhard Schmidt
* [4d56adf8fb] Sean Bright -- res_pjsip_session.c: Prevent
use-after-free with TEST_FRAMEWORK enabled
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
Reported by: Jeremiah Gadd
* [71f86e78b6] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
on hungup channel
Category: Resources/res_pjsip_t38
ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received
Reported by: Salah Ahmed
* [4ac0299bfb] Salah Ahmed -- res_pjsip_t38: T.38 error correction mode
selection at 200 ok received
Category: Resources/res_rtp_asterisk
ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet
length doesn't match
Reported by: Joshua Elson
* [0c486e7edf] Joshua Colp -- res_rtp_asterisk: Always return provided
DTLS packet length.
Category: Resources/res_stasis
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
* [7202624b3b] George Joseph -- stasis: Don't hold app_registry and
session locks unnecessarily
Category: pjproject/pjsip
ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
Reported by: Niklas Larsson
* [2652bda3a0] George Joseph -- pjproject_bundled: Replace earlier
reverts with official fixes.
ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters,
NEC only supports up to 32 characters
Reported by: Dan Cropp
* [a1d38e19a2] Dan Cropp -- pjproject: Configurable setting for cnonce
to include hyphens or not
New Feature
Category: Applications/app_senddtmf
ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
of only "sending"
Reported by: lvl
* [6345a00228] lvl -- app_senddtmf: Add receive mode to AMI Action
PlayDTMF
Category: Core/Jitterbuffer
ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
Reported by: Joshua C. Colp
* [926053d7bd] Joshua Colp -- func_jitterbuffer: Add audio/video sync
support.
Category: Functions/func_curl
ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
Reported by: Martin Tomec
* [d579ec9cdf] Martin Tomec -- func_curl.c: Support custom http headers
Category: Resources/res_musiconhold
ASTERISK-17808: [patch] Unregister a realtime moh class
Reported by: Byron Clark
* [9e26136ee6] sungtae kim -- res_musiconhold: Added unregister realtime
moh class
Category: pjproject/pjsip
ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup
From header URI domain
Reported by: Stas Kobzar
* [3a246c2a69] Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN
----------------------------------------------------------------------
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: Functions/General
ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
Reported by: Pascal Cadotte Michaud
* [450173a0ae] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
argument documentation
----------------------------------------------------------------------
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 |
|------------+----------------------+------------------------------------|
| 08a6e8c553 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 17.1.0 |
|------------+----------------------+------------------------------------|
| 6a89784b78 | Joshua Colp | Revert "PJSIP_CONTACT: add missing |
| | | argument documentation" |
|------------+----------------------+------------------------------------|
| | | res_pjsip_registrar.c: Prevent |
| f26e5bacc0 | Sean Bright | possible buffer overflow with |
| | | domain aliases |
|------------+----------------------+------------------------------------|
| | | channel.c: Resolve issue with |
| 88150323a2 | Thomas Arimont | receiving SIP INFO packets for |
| | | DTMF |
|------------+----------------------+------------------------------------|
| 5b15a1c639 | George Joseph | CI: Turn off shallow cloning |
| | | altogether |
|------------+----------------------+------------------------------------|
| cc59e21409 | Sean Bright | media_cache.c: Various CLI |
| | | improvements |
|------------+----------------------+------------------------------------|
| 2a92e6b576 | George Joseph | CI: Fix missing script block in |
| | | jenkinsfiles |
|------------+----------------------+------------------------------------|
| f0d1ce50af | George Joseph | CI: Fix missing script block in |
| | | jenkinsfiles |
|------------+----------------------+------------------------------------|
| 46dceab33f | George Joseph | CI: Increase clone depth and do |
| | | better cleanup |
|------------+----------------------+------------------------------------|
| 76ef36fafc | Sean Bright | res_pjsip_registrar: Fix |
| | | uninitlized variable warning |
|------------+----------------------+------------------------------------|
| 649733612d | Alexei Gradinari | serializer: set high/low alert |
| | | levels on whole pool |
|------------+----------------------+------------------------------------|
| 8bc6fa0fbd | Kevin Harwell | various files - fix some alerts |
| | | raised by lgtm code analysis |
|------------+----------------------+------------------------------------|
| ea3daa94c8 | Kevin Harwell | res_pjsip_session: initialize |
| | | pending's topology to endpoint's |
|------------+----------------------+------------------------------------|
| | | ExternalMedia: Change return |
| 2d665091a3 | George Joseph | object from ExternalMedia to |
| | | Channel |
|------------+----------------------+------------------------------------|
| b8ae799ca9 | Joshua Colp | res_rtp_asterisk: Remove a log |
| | | message that slipped in. |
|------------+----------------------+------------------------------------|
| ba688e6891 | Joshua Colp | test_res_rtp: Enable FIR and REMB |
| | | nominal tests. |
|------------+----------------------+------------------------------------|
| c84135d2a3 | Chris Savinovich | test_taskprocessor.c: Fix test |
| | | failure on Ubuntu |
|------------+----------------------+------------------------------------|
| 37ec88c4c8 | Kevin Harwell | serializer: move/add asterisk |
| | | serializer pool functionality |
|------------+----------------------+------------------------------------|
| 299ba78b09 | Kevin Harwell | res_pjsip/res_pjsip_mwi: use |
| | | centralized serializer pools |
|------------+----------------------+------------------------------------|
| 25fbe79793 | Corey Farrell | stasis_state: Create internal |
| | | stasis_state_proxy object. |
|------------+----------------------+------------------------------------|
| 4b47d4774d | Alexei Gradinari | res_pjsip_pubsub: add endpoint to |
| | | some warning |
|------------+----------------------+------------------------------------|
| d223419bcd | Jonathan Rose | basic-pbx: Bring forward queue |
| | | configuration from 13 |
|------------+----------------------+------------------------------------|
| | | taskprocessor.c: Added "like" |
| 8269fcbf03 | Ben Ford | support to 'core show |
| | | taskprocessors' |
|------------+----------------------+------------------------------------|
| 37139e16a5 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 17.0.0-rc2 |
|------------+----------------------+------------------------------------|
| 7550a82fe0 | Sean Bright | res_musiconhold: Add new |
| | | 'playlist' mode |
|------------+----------------------+------------------------------------|
| 175a7ccac7 | Kevin Harwell | res_pjsip_pubsub: change warning |
| | | to debug |
|------------+----------------------+------------------------------------|
| 5bedd4a9b4 | Corey Farrell | core: Fix ABI mismatch of |
| | | ao2_global_obj. |
|------------+----------------------+------------------------------------|
| f181dfc065 | Ben Ford | taskprocessor.c: Add CLI commands |
| | | to reset taskprocessor stats. |
|------------+----------------------+------------------------------------|
| ec712cfab2 | Corey Farrell | core: Add |
| | | AO2_ALLOC_OPT_NO_REF_DEBUG option. |
|------------+----------------------+------------------------------------|
| b1c5753b78 | George Joseph | astmm.c: Display backtrace with |
| | | memory show allocations |
|------------+----------------------+------------------------------------|
| af0ccbc45e | Corey Farrell | stasis: refcounter.py can |
| | | incorrectly report skewed objects. |
|------------+----------------------+------------------------------------|
| fd7fc3d7a5 | Corey Farrell | stasis: Fix leaks |
|------------+----------------------+------------------------------------|
| 78f329dad9 | Corey Farrell | app_voicemail: Fix module unload |
| | | leak. |
|------------+----------------------+------------------------------------|
| 1d960195c2 | Ben Ford | res_rtp_asterisk.c: Send RTCP as |
| | | compound packets. |
|------------+----------------------+------------------------------------|
| 5dad4f4ea1 | Ben Ford | res_rtp: Add unit tests for RTCP |
| | | stats. |
|------------+----------------------+------------------------------------|
| 44068cc6bf | George Joseph | ARI: External Media |
|------------+----------------------+------------------------------------|
| c060ef7dee | George Joseph | chan_sip: Update links referenced |
| | | in deprecation notice |
|------------+----------------------+------------------------------------|
| | | test_utils.c: Skip test |
| 7d99710fa6 | Chris-Savinovich | adsi_loaded_test if module not |
| | | loaded. |
|------------+----------------------+------------------------------------|
| | | chan_unistim: Fix code, causing |
| 95b3c9839d | Igor Goncharovsky | all incoming DTMF sent back to |
| | | asterisk |
|------------+----------------------+------------------------------------|
| e7f1f7eefd | Igor Goncharovsky | chan_unistim: Fix RTP port byte |
| | | order for big-endian arch |
|------------+----------------------+------------------------------------|
| a0b0c7540c | Alexei Gradinari | Fix misname 'res_external_mwi' to |
| | | 'res_mwi_external' in comments. |
|------------+----------------------+------------------------------------|
| 581ffdf6a9 | George Joseph | chan_rtp: Accept hostname as well |
| | | as ip address as destination |
|------------+----------------------+------------------------------------|
| b3d9f6ef6c | George Joseph | dns_core: Create new API |
| | | ast_dns_resolve_ipv6_and_ipv4 |
|------------+----------------------+------------------------------------|
| f6ab42df26 | George Joseph | res_ari.c: Prefer exact handler |
| | | match over wildcard |
|------------+----------------------+------------------------------------|
| e123d1ccc5 | Sean Bright | audiohook.c: Substitute silence |
| | | for unavailable audio frames |
|------------+----------------------+------------------------------------|
| 5489ad5574 | George Joseph | CI: Escape backslashes in |
| | | printenv/sort/tr |
|------------+----------------------+------------------------------------|
| aa44e723c5 | George Joseph | CI: Add "throttle" label and |
| | | "skip_gate" capability |
|------------+----------------------+------------------------------------|
| 8652140794 | George Joseph | CI: Make node labels job-specific |
|------------+----------------------+------------------------------------|
| 87f8ca2812 | Sean Bright | app_voicemail: Remove extra |
| | | menuselect build options |
|------------+----------------------+------------------------------------|
| 1976db6ffd | Sean Bright | res_musiconhold: Use a vector |
| | | instead of custom array allocation |
|------------+----------------------+------------------------------------|
| c748187bbf | Sean Bright | manager: Send fewer packets |
|------------+----------------------+------------------------------------|
| f1e0c9d1b8 | Asterisk Development | Update CHANGES and UPGRADE.txt for |
| | Team | 17.0.0 |
|------------+----------------------+------------------------------------|
| 4f2ad5a55d | George Joseph | doc: Add "master-only" flag back |
| | | to the CHANGES and UPGRADE files |
|------------+----------------------+------------------------------------|
| a66ccb90ac | Sean Bright | res_musiconhold: Use |
| | | ast_pipe_nonblock() wrapper |
|------------+----------------------+------------------------------------|
| e821a637a8 | George Joseph | loader.c: Fix possible SEGV when a |
| | | module fails to register |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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 |86152 ----------
asterisk-17.0.0-summary.html | 1265
asterisk-17.0.0-summary.txt | 2973
b/CHANGES | 94
b/README-SERIOUSLY.bestpractices.md | 4
b/UPGRADE.txt | 10
b/addons/cdr_mysql.c | 18
b/addons/chan_ooh323.c | 7
b/apps/app_amd.c | 24
b/apps/app_cdr.c | 8
b/apps/app_chanisavail.c | 3
b/apps/app_dictate.c | 4
b/apps/app_followme.c | 10
b/apps/app_meetme.c | 2
b/apps/app_minivm.c | 3
b/apps/app_mixmonitor.c | 13
b/apps/app_playback.c | 9
b/apps/app_queue.c | 6
b/apps/app_readexten.c | 3
b/apps/app_senddtmf.c | 13
b/apps/app_voicemail.c | 217
b/bridges/bridge_native_rtp.c | 5
b/bridges/bridge_softmix.c | 7
b/cdr/cdr_pgsql.c | 2
b/cel/cel_pgsql.c | 2
b/channels/chan_dahdi.c | 35
b/channels/chan_dahdi.h | 18
b/channels/chan_iax2.c | 16
b/channels/chan_motif.c | 9
b/channels/chan_pjsip.c | 24
b/channels/chan_rtp.c | 19
b/channels/chan_sip.c | 39
b/channels/chan_unistim.c | 174
b/channels/pjsip/cli_commands.c | 13
b/channels/pjsip/dialplan_functions.c | 65
b/channels/pjsip/include/dialplan_functions.h | 25
b/channels/sig_pri.c | 19
b/codecs/Makefile | 3
b/codecs/ex_alaw.h | 5
b/codecs/ex_g722.h | 5
b/codecs/ex_ulaw.h | 5
b/codecs/speex/arch.h | 13
b/codecs/speex/fixed_generic.h | 4
b/codecs/speex/resample.c | 332
b/codecs/speex/speex_resampler.h | 4
b/configs/basic-pbx/extensions.conf | 14
b/configs/basic-pbx/modules.conf | 1
b/configs/basic-pbx/queues.conf | 19
b/configs/samples/extconfig.conf.sample | 1
b/configs/samples/musiconhold.conf.sample | 23
b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py | 54
b/formats/format_g726.c | 16
b/formats/msgsm.h | 4
b/funcs/func_curl.c | 37
b/funcs/func_env.c | 5
b/funcs/func_jitterbuffer.c | 19
b/include/asterisk/abstract_jb.h | 4
b/include/asterisk/ari.h | 2
b/include/asterisk/astobj2.h | 5
b/include/asterisk/audiohook.h | 2
b/include/asterisk/calendar.h | 4
b/include/asterisk/channel_internal.h | 5
b/include/asterisk/config.h | 18
b/include/asterisk/config_options.h | 2
b/include/asterisk/dns_core.h | 22
b/include/asterisk/dns_internal.h | 5
b/include/asterisk/format_cache.h | 5
b/include/asterisk/max_forwards.h | 1
b/include/asterisk/mixmonitor.h | 5
b/include/asterisk/parking.h | 5
b/include/asterisk/res_pjsip.h | 5
b/include/asterisk/res_pjsip_presence_xml.h | 5
b/include/asterisk/res_pjsip_session.h | 2
b/include/asterisk/rtp_engine.h | 111
b/include/asterisk/serializer.h | 85
b/include/asterisk/slin.h | 5
b/include/asterisk/taskprocessor.h | 9
b/include/asterisk/utils.h | 9
b/main/abstract_jb.c | 178
b/main/app.c | 9
b/main/asterisk.c | 15
b/main/astmm.c | 23
b/main/astobj2.c | 88
b/main/astobj2_container.c | 24
b/main/astobj2_global.c | 97
b/main/astobj2_hash.c | 21
b/main/astobj2_rbtree.c | 13
b/main/audiohook.c | 11
b/main/channel.c | 105
b/main/codec_builtin.c | 8
b/main/config.c | 16
b/main/dns_core.c | 72
b/main/event.c | 17
b/main/file.c | 37
b/main/format_cache.c | 8
b/main/indications.c | 6
b/main/manager.c | 92
b/main/media_cache.c | 47
b/main/pbx.c | 5
b/main/pbx_variables.c | 23
b/main/rtp_engine.c | 137
b/main/serializer.c | 189
b/main/stasis.c | 22
b/main/stasis_cache.c | 10
b/main/stasis_state.c | 298
b/main/taskprocessor.c | 219
b/res/ari/config.c | 10
b/res/ari/resource_channels.c | 158
b/res/ari/resource_channels.h | 42
b/res/ari/resource_events.c | 10
b/res/parking/parking_bridge.c | 36
b/res/parking/parking_bridge_features.c | 2
b/res/parking/res_parking.h | 5
b/res/res_ari_channels.c | 135
b/res/res_ari_events.c | 2
b/res/res_calendar_ews.c | 1
b/res/res_calendar_exchange.c | 1
b/res/res_calendar_icalendar.c | 1
b/res/res_config_curl.c | 5
b/res/res_config_pgsql.c | 2
b/res/res_musiconhold.c | 294
b/res/res_phoneprov.c | 6
b/res/res_pjsip.c | 83
b/res/res_pjsip/config_system.c | 2
b/res/res_pjsip/config_transport.c | 17
b/res/res_pjsip/pjsip_configuration.c | 4
b/res/res_pjsip/pjsip_resolver.c | 4
b/res/res_pjsip_endpoint_identifier_ip.c | 18
b/res/res_pjsip_mwi.c | 339
b/res/res_pjsip_outbound_registration.c | 17
b/res/res_pjsip_pubsub.c | 20
b/res/res_pjsip_registrar.c | 55
b/res/res_pjsip_session.c | 19
b/res/res_pjsip_t38.c | 40
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_resolver_unbound.c | 6
b/res/res_rtp_asterisk.c | 276
b/res/res_stasis.c | 43
b/res/stasis/command.c | 2
b/res/stasis/control.c | 2
b/rest-api-templates/res_ari_resource.c.mustache | 2
b/rest-api/api-docs/channels.json | 125
b/tests/CI/gates.jenkinsfile | 12
b/tests/CI/periodics-daily.jenkinsfile | 11
b/tests/CI/ref_debug.jenkinsfile | 9
b/tests/CI/unittests.jenkinsfile | 9
b/tests/test_data_buffer.c | 2
b/tests/test_res_rtp.c | 516
b/tests/test_taskprocessor.c | 28
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1255
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1361
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
157 files changed, 4528 insertions(+), 94603 deletions(-)

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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-17.1.0-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-17.1.0-rc2</h3><h3 align="center">Date: 2019-12-18</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="#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-17.1.0-rc1.</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%">2 Joshua C. Colp <jcolp@sangoma.com><br/>2 George Joseph <gjoseph@digium.com><br/></td><td width="33%"><td width="33%">1 nappsoft <infos@nappsoft.ch><br/>1 Ted G <tgwaste@gmail.com><br/>1 Ted G<br/>1 George Joseph <gjoseph@digium.com><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: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28604">ASTERISK-28604</a>: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d29b06e37de0480ba4c0e3fb61ff690d67572aa">[3d29b06e37]</a> Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28609">ASTERISK-28609</a>: Memory Leak in res_rtp_asterisk.c<br/>Reported by: Ted G<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27a4a3c761613904a6105aec6ff211e206cbbf67">[27a4a3c761]</a> George Joseph -- res_rtp_asterisk: Add frame list cleanups to ast_rtp_read</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5949f9a86a14892ca66d34207f780ed59bff59e5">[5949f9a86a]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5949f9a86a14892ca66d34207f780ed59bff59e5">[5949f9a86a]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</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=a76d3103fb4aa420d9cb7af8cfc5b384d30d6658">a76d3103fb</a></td><td>George Joseph</td><td>Revert "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up"</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>addons/cdr_mysql.c | 8 +--
channels/sig_pri.c | 17 +-------
configure | 82 ++++++++++++++++++++++++++++++++++++---
configure.ac | 20 +++++++++
include/asterisk/autoconfig.h.in | 6 ++
res/res_pjsip_session.c | 20 +++++++++
res/res_rtp_asterisk.c | 2
7 files changed, 129 insertions(+), 26 deletions(-)</pre><br></html>

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Release Summary
asterisk-17.1.0-rc2
Date: 2019-12-18
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Other Changes
5. 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-17.1.0-rc1.
----------------------------------------------------------------------
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
2 Joshua C. Colp 1 nappsoft
2 George Joseph 1 Ted G
1 Ted G
1 George Joseph
----------------------------------------------------------------------
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: Applications/app_meetme
ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
17.0.0
Reported by: George Joseph
* [3d29b06e37] Joshua C. Colp -- configure: Add check for MySQL client
bool and my_bool type usage.
Category: General
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
Reported by: Ted G
* [27a4a3c761] George Joseph -- res_rtp_asterisk: Add frame list
cleanups to ast_rtp_read
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
* [5949f9a86a] Joshua C. Colp -- res_pjsip_session: Set stream state on
created streams for incoming SDP.
Category: Resources/res_pjsip_session
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
* [5949f9a86a] Joshua C. Colp -- res_pjsip_session: Set stream state on
created streams for incoming SDP.
----------------------------------------------------------------------
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 |
|------------+---------------+-------------------------------------------|
| | | Revert "chan_dahdi: set |
| a76d3103fb | George Joseph | CHANNEL(hangupsource) when a PRI channel |
| | | hangs up" |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
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.
addons/cdr_mysql.c | 8 +--
channels/sig_pri.c | 17 +-------
configure | 82 ++++++++++++++++++++++++++++++++++++---
configure.ac | 20 +++++++++
include/asterisk/autoconfig.h.in | 6 ++
res/res_pjsip_session.c | 20 +++++++++
res/res_rtp_asterisk.c | 2
7 files changed, 129 insertions(+), 26 deletions(-)