+++ /dev/null
-<!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-18.0.0</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-18.0.0</h3><h3 align="center">Date: 2020-10-19</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
-<li><a href="#summary">Summary</a></li>
-<li><a href="#contributors">Contributors</a></li>
-<li><a href="#closed_issues">Closed Issues</a></li>
-<li><a href="#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 is the first release of a major new version of Asterisk. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is a new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.</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%">70 Sean Bright <sean.bright@gmail.com><br/>54 George Joseph <gjoseph@digium.com><br/>47 Joshua C. Colp <jcolp@sangoma.com><br/>31 Kevin Harwell <kharwell@digium.com><br/>26 Alexander Traud <pabstraud@compuserve.com><br/>17 Ben Ford <bford@digium.com><br/>16 Joshua Colp <jcolp@digium.com><br/>16 Jaco Kroon <jaco@uls.co.za><br/>11 Walter Doekes <walter+asterisk@wjd.nu><br/>9 sungtae kim <sungtae@messagebird.com><br/>9 Alexander Traud <pabstraud@compuserve.com><br/>8 Torrey Searle <tsearle@voxbone.com><br/>8 Corey Farrell <git@cfware.com><br/>7 Richard Mudgett <rmudgett@digium.com><br/>6 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>6 Pirmin Walthert <infos@nappsoft.ch><br/>6 Guido Falsi <madpilot@FreeBSD.org><br/>5 Asterisk Development Team <asteriskteam@digium.com><br/>5 Alexei Gradinari <alex2grad@gmail.com><br/>3 Pascal Cadotte Michaud <pcm@wazo.io><br/>3 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>3 Jean Aunis <jean.aunis@prescom.fr><br/>3 Nickolay Shmyrev <nshmyrev@alphacephei.com><br/>3 lvl <digium@lvlconsultancy.nl><br/>2 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>2 Andrew Siplas <andrew@asiplas.net><br/>2 Salah Ahmed <txrubel@gmail.com><br/>2 Michael Neuhauser <mike@firmix.at><br/>2 cmaj <chris@penguinpbx.com><br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Michael Goryainov<br/>1 Università di Bologna - CESIA VoIP <cesia.voip@unibo.it><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 Nathan Bruning <nathan@iperity.com><br/>1 Paulo Vicentini <paulo.vicentini@gmail.com><br/>1 Peter Turczak <peter@turczak.de><br/>1 Sungtae Kim <sungtae@messagebird.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Jared Smith <jsmith@fedoraproject.org><br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Daniel Heckl <daniel.heckl@gmail.com><br/>1 Dennis Buteyn <dennis.buteyn@xorcom.com><br/>1 Bernard Merindol <bernard.merindol@telnowedge.com><br/>1 Jonathan Rose <jrose@digium.com><br/>1 Kfir Itzhak <mastertheknife@gmail.com><br/>1 Roger James <roger@beardandsandals.co.uk><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Evandro César Arruda <ecarruda@gmail.com><br/>1 Moises Silva <moises.silva@gmail.com><br/>1 Chris-Savinovich <csavinovich@digium.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Sylvain Afchain <safchain@gmail.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Nicholas John Koch <koch@njk-it.de><br/>1 Peter Sokolov (License #7070)<br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Thomas Arimont (license 5525)<br/>1 Seán C McCord <ulexus@gmail.com><br/>1 Patrick Verzele <patrick@verzele.be><br/>1 snuffy <snuffy22@gmail.com><br/>1 Sebastien Duthil <sduthil@wazo.community><br/>1 Jason Hord <jhord@fluentstream.com> (license 6978)<br/></td><td width="33%">1 tests/test_utils.c.<br/></td><td width="33%">26 Joshua C. Colp <jcolp@digium.com><br/>23 Alexander Traud <pabstraud@compuserve.com><br/>13 Kevin Harwell <kharwell@digium.com><br/>9 Ross Beer <ross.beer@voicehost.co.uk><br/>8 sungtae kim <pchero21@gmail.com><br/>8 nappsoft <infos@nappsoft.ch><br/>7 Walter Doekes <walter+asterisk@wjd.nu><br/>7 Torrey Searle <tsearle@gmail.com><br/>6 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>5 Guido Falsi <madpilot@freebsd.org><br/>5 cmaj <chris@penguinpbx.com><br/>5 George Joseph <gjoseph@digium.com><br/>4 Jaco Kroon <jaco@uls.co.za><br/>4 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>4 Pascal Cadotte Michaud <pascal.cadotte@gmail.com><br/>4 Ross Beer<br/>3 Joshua Elson <joshelson@gmail.com><br/>3 Sean Bright <sean.bright@gmail.com><br/>3 Salah Ahmed <txrubel@gmail.com><br/>3 Nickolay V. Shmyrev <nshmyrev@alphacephei.com><br/>3 lvl <digium@lvlconsultancy.nl><br/>3 Alexei Gradinari <alex2grad@gmail.com><br/>2 Stas Kobzar <stas@modulis.ca><br/>2 Michael Neuhauser <mike@firmix.at><br/>2 Ruddy G <plugworld@micnes.com><br/>2 Joeran Vinzens <vinzens@sipgate.de><br/>2 Timothy Vanderaerden <timothy.vanderaerden@optimise-group.be><br/>2 Sébastien Duthil <sduthil@wazo.community><br/>2 Peter Sokolov <newsletter@fab-online.com><br/>2 Joseph Ades <josephades1@gmail.com><br/>2 Gregory Massel <greg@csurf.co.za><br/>2 Andrew Siplas <andrew@asiplas.net><br/>2 Jared Smith <jaredsmith@jaredsmith.net><br/>2 Jonathan Harris <lardconcepts@gmail.com><br/>2 Michael Neuhauser<br/>1 Ramarajan <pramarajan@sangoma.com><br/>1 Andrey V. T. <avt1203@gmail.com><br/>1 tootai <admin@tootai.net><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 AvayaXAsterisk<br/>1 Joshua C. Colp<br/>1 Etienne Lessard <elessard97@gmail.com><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Yoooooo Ha <n1906374c@e.ntu.edu.sg><br/>1 kevin@phoneburner.com<br/>1 Juan Martin <jmartin79@yandex.com><br/>1 Sylvain Afchain <safchain@wazo.io><br/>1 Speed Dial Dave <speed_dial_dave@gmx.com><br/>1 Andrew Yager <andrew@rwts.com.au><br/>1 Jean-Denis Girard<br/>1 Marian Piater <marian.piater@voipsun.cz><br/>1 Bernard Merindol <bernard.merindol@telnowedge.com><br/>1 Martin Zeh <martin.zeh@forsa.de><br/>1 Corey Farrell <git@cfware.com><br/>1 Dan Cropp <dan@amtelco.com><br/>1 Moises Silva <moises.silva@gmail.com><br/>1 Alexey Vasilyev <alexei.vasilyev@gmail.com><br/>1 Thomas Johnson <tjohnson@microautomation.com><br/>1 Seán C. McCord <ulexus@gmail.com><br/>1 Dirk Wendland<br/>1 Bryan Nelson <bnelson@fluentstream.com><br/>1 Sam Banks <sam.banks.nz@gmail.com><br/>1 Misha Vodsedalek <vmisha@seznam.cz><br/>1 Nicholas John Koch<br/>1 Richard Kenner <kenner@gnat.com><br/>1 EDV O-TON<br/>1 Byron Clark <bclark@getjive.com><br/>1 Christoph Moench-Tegeder <cmt@FreeBSD.org><br/>1 sstream <sstream00@yahoo.co.jp><br/>1 Dmitriy Serov<br/>1 Alex <alex@alex-at.ru><br/>1 candrews <candrews@integralblue.com><br/>1 Sébastien Duthil<br/>1 Robert Sutton <rsutton@noojee.com.au><br/>1 Evandro César Arruda <ecarruda@gmail.com><br/>1 Paul Brooks <paul@dialaround.pro><br/>1 Yury Kirsanov<br/>1 Jason Hord <jhord@fluentstream.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Kevin Flyn<br/>1 Shlomi Gutman <contrib@voicenter.com><br/>1 George Joseph<br/>1 Frank Matano <ftalarico99@gmail.com><br/>1 Cédric Bassaget<br/>1 Dan Jenkins <dan@nimbleape.com><br/>1 Jim Van Meggelen <jim.vanmeggelen@clearlycore.com><br/>1 Patrick Wakano <pwakano@gmail.com><br/>1 Jeremiah Gadd <jeremygadd@gmail.com><br/>1 Michael <ringo@vianet.ca><br/>1 Daniel Heckl <daniel.heckl@gmail.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Kilburn <kilburna@gmail.com><br/>1 Bernhard Schmidt <berni@birkenwald.de><br/>1 Alexander Traud<br/>1 Joeran Vinzens<br/>1 Dennis <dennis.buteyn@xorcom.com><br/>1 Vitold <vit1251@gmail.com><br/>1 Anton Satskiy <satskiy.a@gmail.com><br/>1 Kevin Flyn <kevflynn69@gmail.com><br/>1 David Cunningham <dcunningham@voisonics.com><br/>1 Jim Van Meggelen<br/>1 Vitold<br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Robert Sutton<br/>1 Daniel <depeee@gmail.com><br/>1 Dan Jenkins<br/>1 Ove Aursand <oveaurs@gmail.com><br/>1 Dmitry Wagin <dmitry.wagin@ya.ru><br/>1 Robin Leffmann <robin@stolendata.net><br/>1 Mitch Claborn<br/>1 Jonathan Hunter <jhunter@voxboxcoms.co.uk><br/>1 Joshua Roys <roysjosh@gmail.com><br/>1 Olivier Krief <olivier.krief@gmail.com><br/>1 Paul Brooks<br/>1 Maciej Michno <maciej.michno@xtb.com><br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Niklas Larsson <niklas@tese.se><br/>1 Bernhard Schmidt<br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Maciej Michno<br/>1 Stas Kobzar<br/>1 Cedric BASSAGET <cedric@oceanet.com><br/>1 EDV O-TON <edv@o-ton-online.de><br/>1 Ted G <tgwaste@gmail.com><br/>1 Frank Matano<br/>1 Yury Kirsanov <y.kirsanov@gmail.com><br/>1 Anton Satskiy<br/>1 David M. Lee <dlee@digium.com><br/>1 Patrick Wakano<br/>1 Michael Goryainov <gms4nlt@gmail.com><br/>1 Niklas Larsson<br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Francois Blackburn <fblackburn@wazo.io><br/>1 Università di Bologna - CESIA VoIP <cesia.voip@unibo.it><br/>1 Richard Kenner<br/>1 Niksa Baldun <niksa.baldun@gmail.com><br/>1 Ian Jones <tech@iljones.net><br/>1 Jean-Denis Girard <jd.girard@sysnux.pf><br/>1 Dmitriy Serov <serov.d.p@gmail.com><br/>1 Peter Turczak <peter@turczak.de><br/>1 Roger James <roger@beardandsandals.co.uk><br/>1 Paulo Vicentini <paulo.vicentini@gmail.com><br/>1 Ted G<br/>1 Martin Zeh<br/>1 Università di Bologna - CESIA VoIP<br/>1 Marin Odrljin <marin@maxcom.hr><br/>1 Jonas Swiatek <jonas@telzio.com><br/>1 Eliel Sardañons <eliels@gmail.com><br/>1 AvayaXAsterisk <joh.zuerner@yahoo.de><br/>1 Dirk Wendland <dirk@starface.de><br/>1 Joshua Roys<br/>1 Mark <mark@wrapped.cx><br/>1 Dan Cropp<br/>1 Jonathan Harris<br/>1 Matt Addison <maddison@iquest.net><br/>1 Leandro Dardini <ldardini@gmail.com><br/>1 alex <warp@adygtelecom.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 xrobau <xrobau@gmail.com><br/>1 David Lee<br/>1 Nicholas John Koch <koch@njk-it.de><br/>1 Peter Sokolov<br/>1 Eliel Sardañons<br/>1 Sean Bright<br/>1 Aheliotech <phones@aheliotech.com><br/>1 Bill Kervaski <bill@kervaski.com><br/>1 Cyril Ramière <cyril.ramiere@ino.global><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Niksa Baldun<br/>1 Kfir Itzhak <mastertheknife@gmail.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>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=4a1cadeadb6152e8a7b5959648a9bb4dc7c624f5">[4a1cadeadb]</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=7e3a6e158f965c908dfe1e69f8463a6bab7298f7">[7e3a6e158f]</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=18f5f5fc99a8106c1856d13aeeeeafa2b13ef033">[18f5f5fc99]</a> Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media</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=772b59034ff25b2e943d9e32fdc42e99e7a98609">[772b59034f]</a> lvl -- app_senddtmf: Add receive mode to AMI Action PlayDTMF</li>
-</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-6863">ASTERISK-6863</a>: [patch] allow Asterisk to set high ToS bits as non-root on Linux<br/>Reported by: Matt Addison<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a107e85b2ed03909c821a051b823cc0478e918b6">[a107e85b2e]</a> Alexander Traud -- install_prereq: Add libcap for high bits in DiffServ/ToS.</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=7298a785ad998ac9e5aff36a3d6efa8e4dc82a47">[7298a785ad]</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-17491">ASTERISK-17491</a>: CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything<br/>Reported by: candrews<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c2bf1664c0ce47e68e09846fa7bddcfcfa91e22">[0c2bf1664c]</a> Sean Bright -- func_curl: Add 'followlocation' option to CURLOPT()</li>
-</ul><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=d257a0898eee857f92c6c7f93c13adf1dac21eff">[d257a0898e]</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=cf364cd007c237ccb3a11d40f535b1a3cab7fb29">[cf364cd007]</a> sungtae kim -- res_musiconhold: Added unregister realtime moh class</li>
-</ul><br><h4>Category: Resources/res_pjsip_endpoint_identifier_ip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28639">ASTERISK-28639</a>: res_pjsip_endpoint_identifier_ip: Add ability to match on source port<br/>Reported by: Sean Bright<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=312abaa1fe23967ba95762053310c9c52f0e028a">[312abaa1fe]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add port matching support</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=c7270dca8156194004e4cfcb7bbf5a9a09dbf717">[c7270dca81]</a> Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN</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=172e183b9d1f02ac20ba571de1c6a49a3ad64d32">[172e183b9d]</a> Kevin Harwell -- res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions</li>
-</ul><br><h4>Category: Applications/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28954">ASTERISK-28954</a>: StreamEcho() only returns 1 active stream<br/>Reported by: Bill Kervaski<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=00a52b4752497a6e9e942d70a5e07566a1d6fe9c">[00a52b4752]</a> Joshua C. Colp -- app_stream_echo: Fix state of added streams.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16676">ASTERISK-16676</a>: DAHDIRAS fails to properly initiate pppd unless asterisk is running as root<br/>Reported by: Jaco Kroon<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f92dcd66b5d20b68b1ef991839fc675994bcf8c">[4f92dcd66b]</a> Jaco Kroon -- dahdiras: Only set plugin dahdi.so to pppd if we're running as root.</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=5bda46030049dd61f9dba7758814c1d8e8316c86">[5bda460300]</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-28636">ASTERISK-28636</a>: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.<br/>Reported by: Frederic LE FOLL<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a83625b3668560e9bf70b65961a02eaa52d3739f">[a83625b366]</a> Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.</li>
-</ul><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=2d0eee5418681bc3b55f442552e057923b175e25">[2d0eee5418]</a> Frederic LE FOLL -- ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf.</li>
-</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28841">ASTERISK-28841</a>: app_confbridge: Add support for disabling text messaging for a user<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cfc6ff53cca920e66e3ae6c90837d9215a779f8">[6cfc6ff53c]</a> Joshua C. Colp -- confbridge: Add support for disabling text messaging.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28790">ASTERISK-28790</a>: Crash during conference call using confbridge and video<br/>Reported by: Pascal Cadotte Michaud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=96e8d411e17fc4c07800cde375e485d37cefe05b">[96e8d411e1]</a> Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient space for worst case NACK.</li>
-</ul><br><h4>Category: Applications/app_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28848">ASTERISK-28848</a>: app_fax: Compile.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26b8c999632b24a3e781b692a7f0d7582d180fdd">[26b8c99963]</a> Alexander Traud -- app_fax: SpanDSP headers do not use ast_malloc; ignore that.</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=ed394ce5b1dc62aa35925b64a7c8b90c7634352a">[ed394ce5b1]</a> Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a47cb71bb17d075fdb51634490292baeaa1cbf81">[a47cb71bb1]</a> George Joseph -- Build: Fix compile issues with seldom used modules</li>
-</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28780">ASTERISK-28780</a>: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98d10d0a1612220d1aeaa28c0dcc025844714902">[98d10d0a16]</a> Joshua C. Colp -- audiohook: Don't allow audiohooks to attach to hung up channels.</li>
-</ul><br><h4>Category: Applications/app_osplookup</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28804">ASTERISK-28804</a>: [patch] app_osplookup.c: Avoid a format truncation.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=527e4f6542329a96f0773175352ef9d0e7d70a86">[527e4f6542]</a> Alexander Traud -- app_osplookup: Avoid a format truncation.</li>
-</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25665">ASTERISK-25665</a>: Duplicate logging in queue log for EXITEMPTY events<br/>Reported by: Ove Aursand<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c83e4821e5a334e571383b7b44e0a569cdb9da63">[c83e4821e5]</a> Kfir Itzhak -- app_queue: Fix leave-empty not recording a call as abandoned</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29043">ASTERISK-29043</a>: app_queue: Leave empty sometimes not recorded as abandoned<br/>Reported by: Kfir Itzhak<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c83e4821e5a334e571383b7b44e0a569cdb9da63">[c83e4821e5]</a> Kfir Itzhak -- app_queue: Fix leave-empty not recording a call as abandoned</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29034">ASTERISK-29034</a>: Lastpause of realtime members is reseting<br/>Reported by: Evandro César Arruda<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36dd15c659653e7cfa74914d49b56cecae4eb222">[36dd15c659]</a> Evandro César Arruda -- app_queue: Member lastpause time reseting</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28951">ASTERISK-28951</a>: Inconsistent behaviour queues.conf when there is (not) a [general] section<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=312c23b0e103fe2c240d38d1e760d02af3531d3b">[312c23b0e1]</a> Walter Doekes -- app_queue: (Breaking change) shared_lastcall and autofill default to no</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28950">ASTERISK-28950</a>: Stale code in app_queue to check untouched channel<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=db012e8cc6616d17ff048ea817eb3945b0c13cd4">[db012e8cc6]</a> Walter Doekes -- app_queue: Remove stale code in try_calling</li>
-</ul><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=db012e8cc6616d17ff048ea817eb3945b0c13cd4">[db012e8cc6]</a> Walter Doekes -- app_queue: Remove stale code in try_calling</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0e750cdd10b1488c65175354c012387f41002efa">[0e750cdd10]</a> Walter Doekes -- app_queue: Fix old confusing comment about when the members are called</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28952">ASTERISK-28952</a>: Queue wrapuptime sometimes not respected (based on stale lastcall time)<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fb67383149e06869e66ad2fee0d0c7bab174a37">[0fb6738314]</a> Walter Doekes -- app_queue: Read latest wrapuptime instead of (possibly stale) copy</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28829">ASTERISK-28829</a>: app_queue: leaking stasis subscription when Redirecting call <br/>Reported by: lvl<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f217fcdc62c993e7a8bdb803ab81c4e20765e2e3">[f217fcdc62]</a> Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25844">ASTERISK-25844</a>: app_queue: Ghost channels in "core show channels" output<br/>Reported by: Etienne Lessard<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f217fcdc62c993e7a8bdb803ab81c4e20765e2e3">[f217fcdc62]</a> Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28349">ASTERISK-28349</a>: Pause reason not reported in QueueMember AMI event<br/>Reported by: Niksa Baldun<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9522390a697591cc960a5bc0ffd3d66fcdcb6a25">[9522390a69]</a> Sean Bright -- app_queue: Deprecate the QueueMemberPause.Reason field</li>
-</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28682">ASTERISK-28682</a>: app_record: Lack of `beep` audio file causes application to return error and hangup<br/>Reported by: Corey Farrell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f8b20b949b293b64a24e837109ef935561e5902">[2f8b20b949]</a> Corey Farrell -- app_record: Do not hang up if beep audio is missing</li>
-</ul><br><h4>Category: Applications/app_system</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28776">ASTERISK-28776</a>: Non async-signal-safe syscalls used after fork before exec<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b2d9451741cadb080637190af65d40f692d67ed">[6b2d945174]</a> Pirmin Walthert -- app.c: make sure that no non-async-signal-safe syscalls are used after</li>
-</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27273">ASTERISK-27273</a>: app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command<br/>Reported by: Leandro Dardini<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b575868000b6736ac60b31dfae638bb4d9891f80">[b575868000]</a> Sean Bright -- app_voicemail: Process urgent messages with mailcmd</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23739">ASTERISK-23739</a>: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used<br/>Reported by: Stas Kobzar<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ba8ccb913297c262a2ada47386bb76bc72fa35d2">[ba8ccb9132]</a> Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27622">ASTERISK-27622</a>: empty voicemail.conf required for ARA (realtime) voicemail to leave message<br/>Reported by: Jim Van Meggelen<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9be89d991369c8d750038cd7a320dfee33030665">[9be89d9913]</a> Sean Bright -- app_voicemail: Set globals to default values when voicemail.conf missing</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=15624d9a7af1424480aa9dab217d0c381fa12bd3">[15624d9a7a]</a> Alexei Gradinari -- app_voicemail/IMAP: check mailstream not NULL in leave_voicemail</li>
-</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23739">ASTERISK-23739</a>: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used<br/>Reported by: Stas Kobzar<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ba8ccb913297c262a2ada47386bb76bc72fa35d2">[ba8ccb9132]</a> Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail</li>
-</ul><br><h4>Category: Bridges/bridge_builtin_features</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28920">ASTERISK-28920</a>: bridge show all causes crash<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25ae412f759e49bc0008c5d33f50661851969466">[25ae412f75]</a> sungtae kim -- bridge.c: Fixed null pointer exception</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=7624cbb15578961b5211b84bbbe4efdf5266dc9d">[7624cbb155]</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-28944">ASTERISK-28944</a>: bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8ad06394c4d85684527d41e816d724ef5cad60d7">[8ad06394c4]</a> Joshua C. Colp -- bridge_softmix: Add additional old states for adding new source.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28898">ASTERISK-28898</a>: bridge_softmix: Conference bridge not passing silent rtp packets<br/>Reported by: Jonathan Hunter<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8c8d69d473558043b7f5ffbb7607889e0a619f9">[e8c8d69d47]</a> Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28819">ASTERISK-28819</a>: [patch] bridge_softmix_binaural: Show state in menuselect.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7febd22304721132dc062cca72cd959e99079126">[7febd22304]</a> Alexander Traud -- bridge_softmix_binaural: Show state in menuselect.</li>
-</ul><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=e77cb325831e916fde3fcebf14945ed9864815ed">[e77cb32583]</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-28677">ASTERISK-28677</a>: CDR billsec is always 0 for transferred calls<br/>Reported by: Maciej Michno<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6818c3d1d274dd8da973e693e3d98b565a762818">[6818c3d1d2]</a> George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28636">ASTERISK-28636</a>: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.<br/>Reported by: Frederic LE FOLL<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a83625b3668560e9bf70b65961a02eaa52d3739f">[a83625b366]</a> Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.</li>
-</ul><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=51850a79ef224388173b335d1d2eadd5fed36898">[51850a79ef]</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=52ade18420b346449dff40d6bdb071444cf29e2d">[52ade18420]</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-28702">ASTERISK-28702</a>: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40<br/>Reported by: Andrew Siplas<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5bd72814422623950f0157f4c0fbbabffb6f6601">[5bd7281442]</a> Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"</li>
-</ul><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=a68299f5088cbb1357737bf3e396b69a1cf49fbf">[a68299f508]</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=4072e219f74305e41933350c2b2a2f3aff3bb89a">[4072e219f7]</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=41b67f150eb6464ca501b45c6664be487b5930f3">[41b67f150e]</a> Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up</li>
-</ul><br><h4>Category: Channels/chan_local</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28938">ASTERISK-28938</a>: core_unreal / core_local: Add support for multistream and re-negotiation<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de2813cf235e470c5d3d8ba989826bd860977b79">[de2813cf23]</a> Joshua C. Colp -- core_unreal / core_local: Add multistream and re-negotiation.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25844">ASTERISK-25844</a>: app_queue: Ghost channels in "core show channels" output<br/>Reported by: Etienne Lessard<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f217fcdc62c993e7a8bdb803ab81c4e20765e2e3">[f217fcdc62]</a> Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions</li>
-</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28878">ASTERISK-28878</a>: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16<br/>Reported by: Joseph Ades<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31fbfc5e9511da31e61bafc66d6a4c27feb132df">[31fbfc5e95]</a> Kevin Harwell -- chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4eba6b9eb26f907f1b963db6e4c7ccbf31e1edd1">[4eba6b9eb2]</a> Kevin Harwell -- PJSIP_MEDIA_OFFER: override configuration on refresh</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28886">ASTERISK-28886</a>: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2<br/>Reported by: Jared Smith<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b925fbda3776a9533b4077e059a6fa1565f12ba">[8b925fbda3]</a> Kevin Harwell -- chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28923">ASTERISK-28923</a>: T.38 Segfaults in chan_pjsip_queryoption<br/>Reported by: Yury Kirsanov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41f3a7da4d0ef47ba0b959f019cda3c05b3b8c1c">[41f3a7da4d]</a> George Joseph -- res_fax: Don't start a gateway if either channel is hung up</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28835">ASTERISK-28835</a>: IPv6 addresses in SDP incorrectly formatted<br/>Reported by: Daniel Heckl<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9f117ac9efe560d9a80503a51f2e478bc2edce2a">[9f117ac9ef]</a> Daniel Heckl -- res_pjsip: Fixed format of IPv6 addresses for external media addresses</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28817">ASTERISK-28817</a>: chan_pjsip: constant DTMF tone if RTP is not setup yet<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fa3c8f94e088036836d12a622b7f738d8dd48cc5">[fa3c8f94e0]</a> Kevin Harwell -- chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28774">ASTERISK-28774</a>: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge<br/>Reported by: Michael Neuhauser<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5562fb2ea046c95cf5daaa194bc118742fca7f13">[5562fb2ea0]</a> Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28759">ASTERISK-28759</a>: A non negotiated rtp frame causes call disconnection when there is a SSRC change<br/>Reported by: Paulo Vicentini<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed2a7e3eafd2ead9ed3db5adf3f201fb4fe085dd">[ed2a7e3eaf]</a> Paulo Vicentini -- chan_pjsip: Check audio frame when remote SSRC changes.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28766">ASTERISK-28766</a>: PJSIP blind transfer not completed after using Proceeding()<br/>Reported by: lvl<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1a2ff0aaffb52c91d7b096aeb676a1202d67cdb">[d1a2ff0aaf]</a> lvl -- res_pjsip_refer: ensure refer progress is still sent after Proceeding()</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28755">ASTERISK-28755</a>: SIP/Stasis: SIP headers not transmitted in the "variables" field<br/>Reported by: Jean Aunis - Prescom<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a715cf5aaaaf5af6b2e6496c053e8d21b04f58ea">[a715cf5aaa]</a> Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28492">ASTERISK-28492</a>: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group<br/>Reported by: Jean-Denis Girard<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b40dd11afebb0eae7109b0094ec73447b8e16b1b">[b40dd11afe]</a> Sean Bright -- res_pjsip_config_wizard: Fix change detection for wizard settings</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28502">ASTERISK-28502</a>: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cbc113670474307f5b8bedce35c50f3c056d9c2d">[cbc1136704]</a> George Joseph -- res_pjsip_nat: Restore original contact for REGISTER responses</li>
-</ul><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=ddb0091da5b34e492b4950851618ef7c28c84e9b">[ddb0091da5]</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=bf6f27388d33c7fd8df97d42b2cb6996a525ec63">[bf6f27388d]</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=c03f50c1c87c7eff5ce0f8aa97ceda036ee3781d">[c03f50c1c8]</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=c358da472e73c2cb4d99a6cce4c86d902af7c206">[c358da472e]</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-29011">ASTERISK-29011</a>: chan_sip: ToHost property not cleared on reload<br/>Reported by: Dennis<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9058d9e591e46f26db2f9893355a208407abde40">[9058d9e591]</a> Dennis Buteyn -- chan_sip: Clear ToHost property on peer when changing to dynamic host</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28957">ASTERISK-28957</a>: chan_sip: chan_sip does not process 400 response to an INVITE.<br/>Reported by: Frederic LE FOLL<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a423f935c91788050595f2fd9521b9faf9ad9267">[a423f935c9]</a> Frederic LE FOLL -- chan_sip: chan_sip does not process 400 response to an INVITE.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28898">ASTERISK-28898</a>: bridge_softmix: Conference bridge not passing silent rtp packets<br/>Reported by: Jonathan Hunter<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8c8d69d473558043b7f5ffbb7607889e0a619f9">[e8c8d69d47]</a> Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28651">ASTERISK-28651</a>: chan_sip logs errors on tx to non-existent TCP connections<br/>Reported by: Jaco Kroon<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=365d007eb6f8325bb25e2d20f35cd843422d3dc8">[365d007eb6]</a> Jaco Kroon -- chan_sip: in case of tcp/tls, be less annoying about tx errors.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28647">ASTERISK-28647</a>: chan_sip: RTP frames not transmitted after emitting a COLP<br/>Reported by: Jean Aunis - Prescom<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c9296c635314d671897cd601377cc66982599b0">[9c9296c635]</a> Jean Aunis -- chan_sip: voice frames are no longer transmitted after emitting a COLP</li>
-</ul><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=7624cbb15578961b5211b84bbbe4efdf5266dc9d">[7624cbb155]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
-</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28718">ASTERISK-28718</a>: chan_sip: Returns 403 if RTP ports are depleted, should return 503<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43620cbf6c845624e026d0a8ad72b77f93755e32">[43620cbf6c]</a> Walter Doekes -- chan_sip: Return 503 if we're out of RTP ports</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28686">ASTERISK-28686</a>: chan_sip strictrtp=yes fails when media source is changed: no audio<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=711a3fed564e6eff724d99d8a378bdd8b21aa775">[711a3fed56]</a> Walter Doekes -- chan_sip: Always process updated SDP on media source change</li>
-</ul><br><h4>Category: Channels/chan_sip/Messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28693">ASTERISK-28693</a>: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan<br/>Reported by: Frank Matano<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f309b86e36738d2002c4f48a900e74c22be1a14a">[f309b86e36]</a> Sean Bright -- chan_sip.c: Stop handling continuation lines after reading headers</li>
-</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28372">ASTERISK-28372</a>: Asterisk REPLY Wrong Contact header port (TCP)<br/>Reported by: Anton Satskiy<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52f07176b60fac25164faca24634d5e9b7407a3f">[52f07176b6]</a> Alexander Traud -- chan_sip: externhost/externaddr with non-default TCP/TLS ports.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24428">ASTERISK-24428</a>: Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used<br/>Reported by: sstream<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52f07176b60fac25164faca24634d5e9b7407a3f">[52f07176b6]</a> Alexander Traud -- chan_sip: externhost/externaddr with non-default TCP/TLS ports.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27195">ASTERISK-27195</a>: chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets<br/>Reported by: Joshua Roys<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d0ab620be19f8712b582867153bc85415022d27">[4d0ab620be]</a> Alexander Traud -- chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets.</li>
-</ul><br><h4>Category: Channels/chan_sip/Transfers</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28677">ASTERISK-28677</a>: CDR billsec is always 0 for transferred calls<br/>Reported by: Maciej Michno<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6818c3d1d274dd8da973e693e3d98b565a762818">[6818c3d1d2]</a> George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge</li>
-</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28803">ASTERISK-28803</a>: [patch] chan_unistim: Avoid tautological warnings with clang.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b38f664250531a541f3827e975087aa738e14e6c">[b38f664250]</a> Alexander Traud -- chan_unistim: Avoid tautological warnings with clang.</li>
-</ul><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=3863ab9af9ade586b5c2932842e88325f03583d2">[3863ab9af9]</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=e4289b9e561beab9daa16f69d0dca6ae72747ab5">[e4289b9e56]</a> Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b096389660027365efe893077b24482f84ba31ac">[b096389660]</a> Sean Bright -- codec_resample: Upgrade speex_resample to fix up-sampling bug</li>
-</ul><br><h4>Category: Codecs/codec_silk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28706">ASTERISK-28706</a>: silk 24hHz doesn't show up in 'core show translation' output<br/>Reported by: Sean Bright<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dfad69ce7c43df28bed860a417a4bd06b2987bdd">[dfad69ce7c]</a> Sean Bright -- translate.c: Fix silk 24kHz truncation in 'core show translation'</li>
-</ul><br><h4>Category: Configs/Basic-PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28667">ASTERISK-28667</a>: Asterisk ignores parsing of config files if a Byte order mark is present<br/>Reported by: Robin Leffmann<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40b5cf8f522347a75743f4e6a93ec55cbe9494f7">[40b5cf8f52]</a> Sean Bright -- config.c: Skip UTF-8 BOMs if present when reading config files</li>
-</ul><br><h4>Category: Configs/Samples</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29123">ASTERISK-29123</a>: logger.conf.sample missing comment mark on line 115<br/>Reported by: Andrew Siplas<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79d749d2b5adfc692f5b2640f80a736cdc239aca">[79d749d2b5]</a> Andrew Siplas -- logger.conf.sample: add missing comment mark</li>
-</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27243">ASTERISK-27243</a>: contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax<br/>Reported by: Richard Kenner<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=095c204fe0d3afadf5133036d0f075f31badbd9e">[095c204fe0]</a> snuffy -- contrib/valgrind: Fix use of frame-level suppression</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28664">ASTERISK-28664</a>: "trustrpid" is misspelled in sip_to_pjsip.py<br/>Reported by: Pascal Cadotte Michaud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e494d5fd76988c7642eb7d4fca9f3503a35b6cb9">[e494d5fd76]</a> Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid typo</li>
-</ul><br><h4>Category: Core/ACL</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28978">ASTERISK-28978</a>: acl: named_acl rule misconfiguration results in segfault on reading rule from realtime<br/>Reported by: Andrew Yager<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a43bedd72f8ac7c74033c824103297db9bd56ce">[7a43bedd72]</a> Sean Bright -- acl.c: Coerce a NULL pointer into the empty string</li>
-</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28841">ASTERISK-28841</a>: app_confbridge: Add support for disabling text messaging for a user<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6cfc6ff53cca920e66e3ae6c90837d9215a779f8">[6cfc6ff53c]</a> Joshua C. Colp -- confbridge: Add support for disabling text messaging.</li>
-</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28929">ASTERISK-28929</a>: pjproject_bundled: Honor --without-pjproject.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a4dffe6f85775df4c96223ebdb832c324cfe1a7">[0a4dffe6f8]</a> Alexander Traud -- pjproject_bundled: Honor --without-pjproject.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28837">ASTERISK-28837</a>: pjproject_bundled: Honor --without-pjproject.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=966acc6251bc881f7118276d4dad063abfb24594">[966acc6251]</a> Alexander Traud -- pjproject_bundled: Honor --without-pjproject.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28824">ASTERISK-28824</a>: BuildSystem: Search for Python/C API when possibly needed only.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=610e058189c82e32f85bc5c4dbe4f0b8c483a0ae">[610e058189]</a> Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27717">ASTERISK-27717</a>: [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=610e058189c82e32f85bc5c4dbe4f0b8c483a0ae">[610e058189]</a> Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28816">ASTERISK-28816</a>: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7cdb493a1ede6b3ad77929b3adf3f616bb71becf">[7cdb493a1e]</a> Alexander Traud -- BuildSystem: Remove doc/tex and doc/pdf leftovers.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28818">ASTERISK-28818</a>: [patch] BuildSystem: Allow space in path.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a04947abd961852153e96abff72fd01cd5ec8fb">[7a04947abd]</a> Alexander Traud -- BuildSystem: Allow space in path.</li>
-</ul><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=e40f248faca71a29dcf12cc2849bf41d9c054c22">[e40f248fac]</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-25844">ASTERISK-25844</a>: app_queue: Ghost channels in "core show channels" output<br/>Reported by: Etienne Lessard<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f217fcdc62c993e7a8bdb803ab81c4e20765e2e3">[f217fcdc62]</a> Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28795">ASTERISK-28795</a>: channel: write to a stream on multi-frame writes<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c345ec56d4e8c09ffafa07af91eaf9ba56b2fb7">[3c345ec56d]</a> Kevin Harwell -- channel: write to a stream on multi-frame writes</li>
-</ul><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=1e9714a050109af2bcfa8345cae0ba247638ab6f">[1e9714a050]</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-28955">ASTERISK-28955</a>: "setvar" doesn't work properly in dahdi-channels.conf<br/>Reported by: Marin Odrljin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d88e230037a9810abb69454a8c766492aa16fc5e">[d88e230037]</a> Guido Falsi -- chan_dadhi: Fix setvar in dahdi channels</li>
-</ul><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=32ce6e9a06b5483bf3e6f485ed8c62eae77f4bbf">[32ce6e9a06]</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-28797">ASTERISK-28797</a>: [patch] tcptls: Fix notice when TLS is enabled but not configured.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f9ea75d1179e91322129fcf591ba3d6926e018aa">[f9ea75d117]</a> Alexander Traud -- tcptls: Fix notice when TLS is enabled but not supported.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28839">ASTERISK-28839</a>: Sporadic crashes with Segmentation fault<br/>Reported by: Joeran Vinzens<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e56f4de7e6a51bfbdabb15b41ed1dda920629459">[e56f4de7e6]</a> Joshua C. Colp -- fax: Fix crashes in PJSIP re-negotiation scenarios.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28780">ASTERISK-28780</a>: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98d10d0a1612220d1aeaa28c0dcc025844714902">[98d10d0a16]</a> Joshua C. Colp -- audiohook: Don't allow audiohooks to attach to hung up channels.</li>
-</ul><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=261646c1c4f232ddddd38ca69aa1f3144e09c29b">[261646c1c4]</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=3656c42cb04702e5b223f6984975abae439021ed">[3656c42cb0]</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-28755">ASTERISK-28755</a>: SIP/Stasis: SIP headers not transmitted in the "variables" field<br/>Reported by: Jean Aunis - Prescom<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a715cf5aaaaf5af6b2e6496c053e8d21b04f58ea">[a715cf5aaa]</a> Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI</li>
-</ul><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=729b286d599c8a1c70d40399b9c6010b00d328f8">[729b286d59]</a> Joshua Colp -- stasis: Pass bumped topic_all reference to proxy_dtor.</li>
-</ul><br><h4>Category: Core/Streams</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28870">ASTERISK-28870</a>: streams: One memory leak and one issue cloning streams<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fbfbe7da0b7f82cfbf4bdb1d67ce20c7a3d1884">[7fbfbe7da0]</a> George Joseph -- streams: Fix one memory leak and one formats ref issue</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28846">ASTERISK-28846</a>: stream: Enforce formats immutability<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1c5e68580af4556de86c2c33200fe3d43d499bed">[1c5e68580a]</a> Joshua C. Colp -- stream: Enforce formats immutability and ensure formats exist.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28625">ASTERISK-28625</a>: Playback of local files impacted by large media cache<br/>Reported by: Kevin Reeves<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c626ccec1280b3f65ae14699afb51d62bd16ed99">[c626ccec12]</a> Kevin Reeves -- main/file.c: Limit media cache usage to remote files.</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=084901d5483a6cda7458e9d9f74afeb056f8771d">[084901d548]</a> Torrey Searle -- main/udptl.c: correctly handle udptl sequence wrap around</li>
-</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28816">ASTERISK-28816</a>: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7cdb493a1ede6b3ad77929b3adf3f616bb71becf">[7cdb493a1e]</a> Alexander Traud -- BuildSystem: Remove doc/tex and doc/pdf leftovers.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24484">ASTERISK-24484</a>: Update documentation for statsd module - usage requirements unclear<br/>Reported by: Dan Jenkins<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c376e9f8a824cb6ed71525254ff6de967f7c4294">[c376e9f8a8]</a> Sean Bright -- res_statsd: Document that res_statsd does nothing on its own</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25429">ASTERISK-25429</a>: res_pjsip_endpoint_identifier_ip: Document support for hostnames<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29d867ed67a27ac75c63f4ce83ff1632576a2a52">[29d867ed67]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28507">ASTERISK-28507</a>: Wiki docs missing for MessageWaiting<br/>Reported by: David M. Lee<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5f3ec92d0afc9425ac38cdd4576d0ce7c049256">[d5f3ec92d0]</a> George Joseph -- CI: Update buildAsterisk.sh to do a "make full"</li>
-</ul><br><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=bf4dd3d8374a8f040ec524cfa11b1e404eb95e61">[bf4dd3d837]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7e3015d77913accad9b1dcd200ceec30e52bf445">[7e3015d779]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</li>
-</ul><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=bf7c808604ead43732fcb357fc0c4f6de497b290">[bf7c808604]</a> Sean Bright -- func_env: Prevent FILE() from reading garbage at end-of-file</li>
-</ul><br><h4>Category: Functions/func_aes</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28788">ASTERISK-28788</a>: func_aes: incorrectly printing error 'declined to load'<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cd8cbf7384f9e95737e17ff682ae65f9e06228b4">[cd8cbf7384]</a> Alexander Traud -- func_aes: Avoid incorrect error message on load.</li>
-</ul><br><h4>Category: Functions/func_channel</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28796">ASTERISK-28796</a>: func_channel: cannot read fields exten, context, userfield, channame from dialplan<br/>Reported by: Sébastien Duthil<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d40e34371027224c2b5c1b884df3ea9efab6153e">[d40e343710]</a> Sebastien Duthil -- func_channel: allow reading 4 fields from dialplan</li>
-</ul><br><h4>Category: Functions/func_enum</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26711">ASTERISK-26711</a>: func_enum: ENUM code wrong case<br/>Reported by: Vitold<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=517224ce85dcd0676ffbde5648c82fee84c976e5">[517224ce85]</a> Sean Bright -- enum.c: Add support for regular expression flag in NAPTR record</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19460">ASTERISK-19460</a>: [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab63f0cd0fc9771b65d49894692bde56d321ed68">[ab63f0cd0f]</a> Sean Bright -- enum.c: Make ast_get_txt() actually do something.</li>
-</ul><br><h4>Category: Functions/func_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20325">ASTERISK-20325</a>: Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples.<br/>Reported by: Olivier Krief<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c4e09837421143546b368d3e54a72c25f50e53cd">[c4e0983742]</a> Sean Bright -- func_odbc.conf.sample: Clarify sample documentation</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28497">ASTERISK-28497</a>: func_odbc: truncating Unicode string on readsql<br/>Reported by: Boris P. Korzun<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8979921da949949da11b084a0b6340969a6d828c">[8979921da9]</a> Boris P. Korzun -- func_odbc: acf_odbc_read() and cli_odbc_read() unicode support</li>
-</ul><br><h4>Category: Functions/func_version</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29021">ASTERISK-29021</a>: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions<br/>Reported by: cmaj<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=543f9361474130504acb3ae8d948fc251fe65022">[543f936147]</a> cmaj -- Makefile: Fix certified version numbers</li>
-</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28930">ASTERISK-28930</a>: ./configure --without-ssl build failure<br/>Reported by: Jaco Kroon<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9b5042433be82a7bda803d3a70d6f36fb3769437">[9b5042433b]</a> Joshua C. Colp -- menuselect: Resolve infinite loop in dependency scenario.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28838">ASTERISK-28838</a>: AST_MODULE_INFO requires, MODULEINFO does not mention<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=abf4d74384fdd2ebb92bf8bfc55900b9fc38e31f">[abf4d74384]</a> Alexander Traud -- cdr_odbc: Sync load- and build-time deps.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=191f13626044b318011be476e71a605fb835d00e">[191f136260]</a> Alexander Traud -- res_pjsip_refer: Add build-time dependency.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c2b8fdeca986d812f5a6cd41c3369a675ab41a2">[5c2b8fdeca]</a> Alexander Traud -- app_getcpeid: Add build-time dependency.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=008f46bf1e5822d64d9f192aa2367e9f2fea1a09">[008f46bf1e]</a> Alexander Traud -- res_pjsip: Sync load- and build-time deps.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e2affa3b0a9df59c1d8debf93d25334093d1a04f">[e2affa3b0a]</a> Alexander Traud -- curl: Add build-time dependency.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1135b453ba5a6fbfffba0d11629bc9095161a5a">[f1135b453b]</a> Alexander Traud -- res_pjsip: Add build-time dependency.</li>
-</ul><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=39c920ac78bdc8855c6b005e3d95d828690df023">[39c920ac78]</a> George Joseph -- res_rtp_asterisk: Add frame list cleanups to ast_rtp_read</li>
-</ul><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=a4222614c4d8f0b2bc362b80cb4178fa5d0dcf52">[a4222614c4]</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=a4caaef64c81eaa3b7e8e78ac491d88da61a74e2">[a4caaef64c]</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=b805e1237d7de0ee5ad165510e274288f6a8dadc">[b805e1237d]</a> Kevin Harwell -- srtp: Fix possible race condition, and add NULL checks</li>
-</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29046">ASTERISK-29046</a>: pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension<br/>Reported by: Ramarajan<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d50d152d83f8dc676755b815f127ba539f0b726">[6d50d152d8]</a> Joshua C. Colp -- pbx: Fix hints deadlock between reload and ExtensionState.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28695">ASTERISK-28695</a>: core: minmemfree watermark uses free RAM, not available RAM<br/>Reported by: Kevin Flyn<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50d02d61943fce1de8eef915524f56864d813654">[50d02d6194]</a> Sean Bright -- pbx.c: Include filesystem cache in free memory calculation</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28605">ASTERISK-28605</a>: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X<br/>Reported by: Dirk Wendland<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ee7d72eb72f3a7fc9864e47c3264a353af4a80c8">[ee7d72eb72]</a> George Joseph -- sig_pri: Fix deadlock caused by sig_pri_queue_hangup</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=702019fc80b45216b7ba95f24345d2b0953d3d6b">[702019fc80]</a> Sean Bright -- pbx: Prevent Realtime switch crash on invalid priority</li>
-</ul><br><h4>Category: PBX/pbx_dundi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21205">ASTERISK-21205</a>: [patch] dundi_read_result crash due to negative number<br/>Reported by: Jaco Kroon<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40e93b0240e5f663d372e94306c0b9715aad8a0f">[40e93b0240]</a> Jaco Kroon -- dundi: fix NULL dereference.</li>
-</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28948">ASTERISK-28948</a>: ARI channel create doesn't referencing the channel_id parameter<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbe0f2230d12fd76f1100f47ebc867290dda846d">[bbe0f2230d]</a> sungtae kim -- res_ari: Fix create channel request channelId parameter parsing</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28679">ASTERISK-28679</a>: stasis application is destroyed after its creation<br/>Reported by: Francois Blackburn<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4206830a5269dfc5e8d780655ae641e13eba69d4">[4206830a52]</a> Kevin Harwell -- res_stasis: trigger cleanup after update</li>
-</ul><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=360936ead5a421cf888ec08cd402f2ecddb5c6e7">[360936ead5]</a> Joshua Colp -- res_ari_events: Add module reference when a WebSocket is open.</li>
-</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28898">ASTERISK-28898</a>: bridge_softmix: Conference bridge not passing silent rtp packets<br/>Reported by: Jonathan Hunter<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8c8d69d473558043b7f5ffbb7607889e0a619f9">[e8c8d69d47]</a> Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.</li>
-</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28940">ASTERISK-28940</a>: /channels/create doesn't get any parameters from the body<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fa7c69f40fc34323baa7c71ae2f68b9676a9c62d">[fa7c69f40f]</a> sungtae kim -- res_ari: Fix create request body parameter parsing.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28847">ASTERISK-28847</a>: ARI channels cuts the endpoint string over 80 characters<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ad3d2829c58da0a2f75acae72ba456cafc6f593">[9ad3d2829c]</a> sungtae kim -- res_ari_channels: Fixed endpoint 80 characters limit</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=16e668c7dd8d721beab89600dea08b9a3157049b">[16e668c7dd]</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=16e668c7dd8d721beab89600dea08b9a3157049b">[16e668c7dd]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
-</ul><br><h4>Category: Resources/res_corosync</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28888">ASTERISK-28888</a>: res_corosync: causes asterisk crash in huge distributed environment.<br/>Reported by: Università di Bologna - CESIA VoIP<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c1c386634530ad6216058627f8ea70ff3e74b3e">[0c1c386634]</a> Università di Bologna - CESIA VoIP -- res_corosync: Fix crash in huge distributed environment.</li>
-</ul><br><h4>Category: Resources/res_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28900">ASTERISK-28900</a>: res_fax: Double frame free when gateway in use with off-nominal format usage<br/>Reported by: Gregory Massel<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2500c6273ec4eb59267da87e8268295b34af246">[d2500c6273]</a> Joshua C. Colp -- res_fax: Don't consume frames given to fax gateway on write.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28660">ASTERISK-28660</a>: res_fax: wrap Asterisk initiated negotiation with config option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b6f56073595800e1bd0b8deafe8259a9da73e07a">[b6f5607359]</a> Kevin Harwell -- res_fax: wrap v21 detected Asterisk initiated negotiation with config option</li>
-</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28975">ASTERISK-28975</a>: res_http_websocket: Text payload data doesn't necessary include trailing zero<br/>Reported by: Nickolay V. Shmyrev<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4d24f513734fbb2cff49c355b475851b4c701c3">[e4d24f5137]</a> Nickolay Shmyrev -- res_http_websocket: Avoid reading past end of string</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28562">ASTERISK-28562</a>: SIP WSS message not processed until next frame arrives<br/>Reported by: Robert Sutton<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87110c1bdf9a20c44a4109e9475f6278c8f56d50">[87110c1bdf]</a> Sean Bright -- websocket: Consider pending SSL data when waiting for socket input</li>
-</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28927">ASTERISK-28927</a>: Asterisk crash in music on hold<br/>Reported by: David Cunningham<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=57554c28343e8df02bdc6ea115f09cd603c6b78e">[57554c2834]</a> Sean Bright -- res_musiconhold.c: Prevent crash with realtime MoH</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28892">ASTERISK-28892</a>: res_musiconhold: Module res_musiconhold throws false warning<br/>Reported by: Nicholas John Koch<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fef97a9a72e19c0825074de8f5e6433dd10007ea">[fef97a9a72]</a> Nicholas John Koch -- res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28735">ASTERISK-28735</a>: Realtime MoH Unknown format '' -- defaulting to SLIN<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeff1f2c530928d00bf7028e43a6bb7f5533703e">[aeff1f2c53]</a> Sean Bright -- res_musiconhold: Avoid spurious warning when 'format' is the empty string</li>
-</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29042">ASTERISK-29042</a>: res_parking: Parker UUID is no longer copied<br/>Reported by: Misha Vodsedalek<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f0766dcda575269608497c59985d726f52a7981">[4f0766dcda]</a> Joshua C. Colp -- parking: Copy parker UUID as well.</li>
-</ul><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=811ae88da4c6e4b6c5360fdf60a825d844311839">[811ae88da4]</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=807a70b7ae1c18e7d66eca517647ca31efa73bb3">[807a70b7ae]</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=e924c5107c7bd27dfb63d8e14beb666224356773">[e924c5107c]</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-28995">ASTERISK-28995</a>: res_pjsip_registrar: Expires on statically configured contacts is not correct<br/>Reported by: tootai<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99eafe5771a3c49f66da0be1710acb954ca83eb3">[99eafe5771]</a> Joshua C. Colp -- res_pjsip_registrar: Don't specify an expiration for static contacts.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28965">ASTERISK-28965</a>: res_pjsip: Apply outbound proxy to static contacts on AOR<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f86118bd8fb4d31de79b97cbe57963e73fdd336">[4f86118bd8]</a> Joshua C. Colp -- res_pjsip: Apply AOR outbound proxy to static contacts.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28936">ASTERISK-28936</a>: res_pjsip: crash when dialing non-sip uri<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e74dde5100e5ff2d6531a9150b37584fb22d088f">[e74dde5100]</a> Walter Doekes -- pjsip: Prevent invalid memory access when attempting to contact a non-sip URI</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28794">ASTERISK-28794</a>: res_pjsip: Crash when escaping during URI printing<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c2871edf4df413fdb43dbaee063677b970f21a2">[9c2871edf4]</a> Joshua C. Colp -- res_pjsip: Use correct pool for storing the contact_user value.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26780">ASTERISK-26780</a>: res_pjsip: PJSIP Registration Fails when transport=transport-udp6<br/>Reported by: Peter Sokolov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8dec423d219438b735e34987e37718838586fa2">[c8dec423d2]</a> Peter Sokolov -- pjsip_resolver.c: Ensure AAAA dns requests are made.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28854">ASTERISK-28854</a>: SIGSEGV when pjsip show history encounters IPV6 address<br/>Reported by: Roger James<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4a072c48900bf1166582cefd141914153bbd51ab">[4a072c4890]</a> Roger James -- res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28056">ASTERISK-28056</a>: res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR<br/>Reported by: Jason Hord<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d845464c76ae27c94c25130a21372cdb42b6db1d">[d845464c76]</a> Jason Hord -- res_pjsip: Don't set endpoint to unavailable in all cases.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28790">ASTERISK-28790</a>: Crash during conference call using confbridge and video<br/>Reported by: Pascal Cadotte Michaud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=96e8d411e17fc4c07800cde375e485d37cefe05b">[96e8d411e1]</a> Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient space for worst case NACK.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28743">ASTERISK-28743</a>: Asterisk is crashing if the 200 OK with SDP<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8147f437569bdb1841d4f326675a5b84dbb078e1">[8147f43756]</a> Sungtae Kim -- res_pjsip_session: Fixed wrong session termination</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23407">ASTERISK-23407</a>: Fix the FSF address in the headers of lots of pjproject files<br/>Reported by: Jared Smith<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a7fe3097f1e92db5e77ca1a7a34411e30bc913d">[0a7fe3097f]</a> Jared Smith -- indications.conf.sample: Add indication tones for Indonesia</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28139">ASTERISK-28139</a>: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls<br/>Reported by: Paul Brooks<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=313189aae22cf3f662d4b024fa57bec1bf4ddfd1">[313189aae2]</a> Sean Bright -- chan_pjsip: Ignore RTP that we haven't negotiated</li>
-</ul><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=b1be06df8dc898a863623fd8fa4662bb9926fe28">[b1be06df8d]</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=377d7bdab69ff4f295222949f15d8880e43ad0c3">[377d7bdab6]</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=cc83e76aa5f5277d4c8dc7e94acf39cfc9072a6d">[cc83e76aa5]</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=86452c9fa4cf013db041322eb58339f90ac7700f">[86452c9fa4]</a> Joshua Colp -- res_pjsip: Fix multiple of the same contact in "pjsip show contacts".</li>
-</ul><br><h4>Category: Resources/res_pjsip_acl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28697">ASTERISK-28697</a>: res_pjsip: Named ACL does not update on reload if changed<br/>Reported by: Timothy Vanderaerden<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d6712790cd49cf3050f2a3783efc07fc5396c0ad">[d6712790cd]</a> Joshua C. Colp -- pjsip: Update ACLs on named ACL changes.</li>
-</ul><br><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29001">ASTERISK-29001</a>: chan_pjsip does not process or forward 181 responses<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=addd295cdacd0dc93b0084b688acfaf43d13d653">[addd295cda]</a> Torrey Searle -- res_pjsip_diversion: handle 181</li>
-</ul><br><h4>Category: Resources/res_pjsip_endpoint_identifier_ip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25429">ASTERISK-25429</a>: res_pjsip_endpoint_identifier_ip: Document support for hostnames<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29d867ed67a27ac75c63f4ce83ff1632576a2a52">[29d867ed67]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames</li>
-</ul><br><h4>Category: Resources/res_pjsip_logger</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28932">ASTERISK-28932</a>: res_pjsip_logger writing too big packets<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8c6e9ae5ded4f87302970a87f916ed8a3a6462f">[e8c6e9ae5d]</a> Pirmin Walthert -- res_pjsip_logger: use the correct pointer when logging tx_messages to pcap</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28921">ASTERISK-28921</a>: Wrong return value check for fwrite when writing to pcap file<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c16937cdbea44e85cf32d5b849d873427d501a43">[c16937cdbe]</a> Pirmin Walthert -- res_pjsip_logger.c: correct the return value checks when writing to pcap</li>
-</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26082">ASTERISK-26082</a>: res_pjsip_messaging: MessageSend Content-Type can't be changed<br/>Reported by: Alex<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03d24ca4c144e3506c03af75df9e764084b5a51e">[03d24ca4c1]</a> Sean Bright -- res_pjsip_messaging: Allow Content-Type to be overridden</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25421">ASTERISK-25421</a>: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending<br/>Reported by: Dmitriy Serov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1ca2c5d71e028388a45f96d06834a88f1a20974">[b1ca2c5d71]</a> Sean Bright -- res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly</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=5dae803eea01a0574eb34bc92933fabc69504767">[5dae803eea]</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=12dbeb69b093d9eab22848e5b8122b49d20c6f17">[12dbeb69b0]</a> Kevin Harwell -- res_pjsip_mwi: use an ao2_global object for mwi containers</li>
-</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28884">ASTERISK-28884</a>: x-ast-orig-host not filtered out from request URI and To header<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1399f8b4fe980286a0ca9e84eeaea972474da50d">[1399f8b4fe]</a> Pirmin Walthert -- res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header</li>
-</ul><br><h4>Category: Resources/res_pjsip_notify</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27775">ASTERISK-27775</a>: res_pjsip_notify: Multiple Event headers can be present instead of just one<br/>Reported by: AvayaXAsterisk<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90af050fa4b010934b1b17e894c583b0c9076c73">[90af050fa4]</a> Sean Bright -- res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY</li>
-</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28746">ASTERISK-28746</a>: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78b01f41aec96035f8f030934ce8db0a78f4a8f6">[78b01f41ae]</a> George Joseph -- res_pjsip_outbound_registration: Fix SRV failover on timeout</li>
-</ul><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=d5d41409e245c964088cd88798ef15a21a3ff7de">[d5d41409e2]</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=cc83e76aa5f5277d4c8dc7e94acf39cfc9072a6d">[cc83e76aa5]</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=982a5025b3121596072839195cf676d1ff88ed63">[982a5025b3]</a> Sean Bright -- res_pjsip_registrar: Validate Contact URI before adding to responses</li>
-</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28714">ASTERISK-28714</a>: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1f0c833abf927a7a083171526e0609884e676d4">[a1f0c833ab]</a> Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27759">ASTERISK-27759</a>: res_pjsip_pubsub: Subscription persistence does not preserve XML <dialog-info> version number<br/>Reported by: Bryan Nelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e7adbd8f4c416a9f5a32fba5148b038c4f40b37">[4e7adbd8f4]</a> Joshua C. Colp -- res_pjsip_pubsub: Add ability to persist generator state information.</li>
-</ul><br><h4>Category: Resources/res_pjsip_registrar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28995">ASTERISK-28995</a>: res_pjsip_registrar: Expires on statically configured contacts is not correct<br/>Reported by: tootai<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99eafe5771a3c49f66da0be1710acb954ca83eb3">[99eafe5771]</a> Joshua C. Colp -- res_pjsip_registrar: Don't specify an expiration for static contacts.</li>
-</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28784">ASTERISK-28784</a>: res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34750d2068c6b3c230d6fa0762e553ee1d7b28ed">[34750d2068]</a> Joshua C. Colp -- res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28774">ASTERISK-28774</a>: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge<br/>Reported by: Michael Neuhauser<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5562fb2ea046c95cf5daaa194bc118742fca7f13">[5562fb2ea0]</a> Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28754">ASTERISK-28754</a>: ASTERISK-28738 Causes Audio Issue After Hold<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77c9ba8e635eac24264138fc27e3e652e2eb15df">[77c9ba8e63]</a> Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH transitions</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28738">ASTERISK-28738</a>: Incorrect state machine used when MOH_PASSTHRU is used<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bf4340f0eceb9833a62ddf16aecfd3ccd7f1dc6e">[bf4340f0ec]</a> Torrey Searle -- res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough</li>
-</ul><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=a603d7d324f7a905cb42dd45f608e4708488e057">[a603d7d324]</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-29109">ASTERISK-29109</a>: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5cc4a391b3db21198178093f0c65550d5eabb7ec">[5cc4a391b3]</a> Joshua C. Colp -- res_pjsip: Adjust outgoing offer call pref.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29033">ASTERISK-29033</a>: res_pjsip_session: Aggressively terminates session on failed re-INVITE<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c074038fee984fd8e83ca8f48337d00f802e179">[3c074038fe]</a> Joshua C. Colp -- res_pjsip_session: Don't aggressively terminate on failed re-INVITE.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28953">ASTERISK-28953</a>: res_pjsip_session: Preserve stream label<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ee8ea9275ffb0adf20faae2e8fce4bbad330e994">[ee8ea9275f]</a> Joshua C. Colp -- res_pjsip_session: Preserve label on incoming re-INVITE.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28871">ASTERISK-28871</a>: res_pjsip_session: Unnecessary re-Invite on call answer<br/>Reported by: Alexei Gradinari<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afa2c9a868d919824f9df410548a0f8863c56680">[afa2c9a868]</a> Joshua C. Colp -- bridge: Don't try to match audio formats.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28783">ASTERISK-28783</a>: res_pjsip_session: Allow default non-audio streams to have reflected state<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9620ecbf80fa8a3b93f22798489e3d0bb469dfab">[9620ecbf80]</a> Joshua C. Colp -- res_pjsip_session: Don't restrict non-audio default streams to sendrecv.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28730">ASTERISK-28730</a>: res_pjsip_session: Fix out of order session refreshes<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac155decaed82a36f8ee9aed5811ef0180ca5cbe">[ac155decae]</a> Joshua C. Colp -- res_pjsip_session: Fix off-nominal session refreshes.</li>
-</ul><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=a603d7d324f7a905cb42dd45f608e4708488e057">[a603d7d324]</a> Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.</li>
-</ul><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=6ee1f1f50741927f333ed057d18339dda2b02d78">[6ee1f1f507]</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=c03f50c1c87c7eff5ce0f8aa97ceda036ee3781d">[c03f50c1c8]</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=330ffa2bcea8c1c3c1ae2bca9704718cbc08c7cb">[330ffa2bce]</a> Salah Ahmed -- res_pjsip_t38: T.38 error correction mode selection at 200 ok received</li>
-</ul><br><h4>Category: Resources/res_realtime</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21794">ASTERISK-21794</a>: CLI command 'realtime update2' syntax failure when using according to usage help<br/>Reported by: Cedric BASSAGET<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=094e87b0dcaa8d7dde4539b5d2299fe96da4be7c">[094e87b0dc]</a> Sean Bright -- res_realtime: Fix 'realtime update2' argument handling</li>
-</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28974">ASTERISK-28974</a>: res_rtp_asterisk: T.140 messages have appended RTP string to each message block.<br/>Reported by: Thomas Johnson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5ec7099312ae5b9beff8cedb3e02acb6026f9fa0">[5ec7099312]</a> Sean Bright -- bridge_channel: Ensure text messages are zero terminated</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28939">ASTERISK-28939</a>: res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c84d962eae394946b777c769522dc9541f1e4540">[c84d962eae]</a> Joshua C. Colp -- res_rtp_asterisk: Don't assume setting retrans props means to enable.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28904">ASTERISK-28904</a>: RTP ICE leaks the memory<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8c94b6cf1861d2106e7199ebc4de5b894dd8661">[c8c94b6cf1]</a> sungtae kim -- res_rtp_asterisk.c: Fixed memory leak</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28852">ASTERISK-28852</a>: Unprotected access to nochecksums variable, causes build failures<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4366308e192a521aaf8f0d3da0c625308782f5e">[e4366308e1]</a> Guido Falsi -- res_rtp_asterisk: Protect access to nochecksums with #ifdef</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28827">ASTERISK-28827</a>: res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d50fd0acc00dcc4ffdcc48e27375752b842957dd">[d50fd0acc0]</a> Pirmin Walthert -- res_rtp_asterisk: Resolve loop when receive buffer is flushed</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28826">ASTERISK-28826</a>: res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ca032d1e2e31ff865f8e9acb0f80769b868dd1a8">[ca032d1e2e]</a> Pirmin Walthert -- res_rtp_asterisk: Free payload when error on insertion to data buffer</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28812">ASTERISK-28812</a>: First DTMF is not get<br/>Reported by: Bernard Merindol<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7db03e12a7448e6b0289ad0b62081c1d52886b50">[7db03e12a7]</a> Bernard Merindol -- res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28809">ASTERISK-28809</a>: [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ef1b1b0c2df9c4e8251f8854968863715cb64c8">[1ef1b1b0c2]</a> Alexander Traud -- res_rtp_asterisk: Avoid absolute value on unsigned subtraction.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28773">ASTERISK-28773</a>: Incorrect Sender SSRC in RTCP when p2p rtp bridge is active<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1dba820cf96fbc8cd5f30760f5077f5b1fe3e8b">[a1dba820cf]</a> Torrey Searle -- res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28769">ASTERISK-28769</a>: DTLS Handshake Fails to Occur if ice_support is enabled but not used<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14ba1806f382d616cdbf58dfa643431d82537a3a">[14ba1806f3]</a> Torrey Searle -- res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28759">ASTERISK-28759</a>: A non negotiated rtp frame causes call disconnection when there is a SSRC change<br/>Reported by: Paulo Vicentini<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed2a7e3eafd2ead9ed3db5adf3f201fb4fe085dd">[ed2a7e3eaf]</a> Paulo Vicentini -- chan_pjsip: Check audio frame when remote SSRC changes.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28764">ASTERISK-28764</a>: res_rtp_asterisk: Improve NACK support and seqno handling<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87fda066eac730df655bfbf2527d6cd449cd832d">[87fda066ea]</a> Joshua C. Colp -- res_rtp_asterisk: Improve video performance in certain networks.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28716">ASTERISK-28716</a>: ICE: pjnath shouldn't wait for ICE to complete before allowing sending<br/>Reported by: Benjamin Keith Ford<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=168637cc0c19c9ffdf7c43a8a4e2e07a89b503b1">[168637cc0c]</a> Ben Ford -- RTP/ICE: Send on first valid pair.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28742">ASTERISK-28742</a>: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3865b3fd6a2e8c98158aee0713d81913bf84ca06">[3865b3fd6a]</a> Kevin Harwell -- res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup</li>
-</ul><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=02129ad4d08724ba0dcc89303c835220992a4bb3">[02129ad4d0]</a> Joshua Colp -- res_rtp_asterisk: Always return provided DTLS packet length.</li>
-</ul><br><h4>Category: Resources/res_sorcery_memory_cache</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28942">ASTERISK-28942</a>: res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a143c3a7b77ca1928530ea683f76ec4956a0a59c">[a143c3a7b7]</a> Joshua C. Colp -- res_sorcery_memory_cache: Disallow per-object expire with full backend.</li>
-</ul><br><h4>Category: Resources/res_speech</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29040">ASTERISK-29040</a>: res_speech: Assertion on format<br/>Reported by: Nickolay V. Shmyrev<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0319e0b07f7a38860575254cfd4188682cd89eb1">[0319e0b07f]</a> Nickolay Shmyrev -- res_speech: Bump reference on format object</li>
-</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28903">ASTERISK-28903</a>: res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4de0e50c320a89ae0938008642716ac19802faf0">[4de0e50c32]</a> Alexander Traud -- res_srtp: Set all possible flags while selecting the Crypto Suite.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22920">ASTERISK-22920</a>: Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling<br/>Reported by: Shlomi Gutman<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29070b61f74cda54bacf60011d51aa92a5b805f2">[29070b61f7]</a> Alexander Traud -- core_local: Local calls are always secure.</li>
-</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28987">ASTERISK-28987</a>: BridgeCreated ARI event shows wrong video_mode info<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2e32b56bdb5d9570b7344943b27212038a6e4fbe">[2e32b56bdb]</a> sungtae kim -- stasis_bridge.c: Fixed wrong video_mode shown</li>
-</ul><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=cce2b0da954fb232cafc812ce5acd6f3b42341e9">[cce2b0da95]</a> Kevin Harwell -- stasis/app: don't lock an app before a call to send</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=990a91b44ae055d003ebd4890ed2c74c5e0636a0">[990a91b44a]</a> George Joseph -- stasis: Don't hold app_registry and session locks unnecessarily</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28633">ASTERISK-28633</a>: stasis bridge topic leak<br/>Reported by: Joeran Vinzens<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1c9ddad4db326ad04e86bd17ef8b5d1fe5a8006d">[1c9ddad4db]</a> George Joseph -- stasis.c: Use correct topic name in stasis_topic_pool_delete_topic</li>
-</ul><br><h4>Category: Resources/res_stasis_playback</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28713">ASTERISK-28713</a>: res_stasis_playback: Error building JSON<br/>Reported by: Sébastien Duthil<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31dc9043801039647903b3bff44d7988e056dcc2">[31dc904380]</a> Sean Bright -- res_stasis_playback: Prevent media_index from going out of bounds</li>
-</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24484">ASTERISK-24484</a>: Update documentation for statsd module - usage requirements unclear<br/>Reported by: Dan Jenkins<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c376e9f8a824cb6ed71525254ff6de967f7c4294">[c376e9f8a8]</a> Sean Bright -- res_statsd: Document that res_statsd does nothing on its own</li>
-</ul><br><h4>Category: Tests/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28808">ASTERISK-28808</a>: [patch] test_stasis: Avoid always true warning with clang.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb28ed0d1bdbd0480616f738c2c216f46d9075cc">[bb28ed0d1b]</a> Alexander Traud -- test_stasis: Avoid always true warning with clang.</li>
-</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27717">ASTERISK-27717</a>: [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=610e058189c82e32f85bc5c4dbe4f0b8c483a0ae">[610e058189]</a> Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28789">ASTERISK-28789</a>: test_utils: incorrectly printing error 'declined to load'<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc07eeaba16077cc5c8fcfeafc928f683127286f">[fc07eeaba1]</a> Alexander Traud -- test_utils: Avoid incorrect error message on load.</li>
-</ul><br><h4>Category: Utilities/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28685">ASTERISK-28685</a>: check_expr2: linking (when hardening) and cross-compiling troubles<br/>Reported by: Sebastian Kemper<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b7fbb9c41f3c1a071c594c68ff0dcc763291d451">[b7fbb9c41f]</a> Sebastian Kemper -- check_expr2: fix cross-compile/hardening issues</li>
-</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28973">ASTERISK-28973</a>: Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address)<br/>Reported by: Michael Neuhauser<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6482ab5bea791fe94d6f0f6b617d28eb0256a36c">[6482ab5bea]</a> Michael Neuhauser -- pjproject: clone sdp to protect against (nat) modifications</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28929">ASTERISK-28929</a>: pjproject_bundled: Honor --without-pjproject.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a4dffe6f85775df4c96223ebdb832c324cfe1a7">[0a4dffe6f8]</a> Alexander Traud -- pjproject_bundled: Honor --without-pjproject.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28794">ASTERISK-28794</a>: res_pjsip: Crash when escaping during URI printing<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c2871edf4df413fdb43dbaee063677b970f21a2">[9c2871edf4]</a> Joshua C. Colp -- res_pjsip: Use correct pool for storing the contact_user value.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28859">ASTERISK-28859</a>: pjsip: Increase maximum candidate count<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3078a00a6dbe7e1e7f476c5d49daf5cff58d8b38">[3078a00a6d]</a> Joshua C. Colp -- pjsip: Increase maximum ICE candidate count.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28811">ASTERISK-28811</a>: Crash occurs when fax session switches from T.38 to audio<br/>Reported by: Alexey Vasilyev<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e56f4de7e6a51bfbdabb15b41ed1dda920629459">[e56f4de7e6]</a> Joshua C. Colp -- fax: Fix crashes in PJSIP re-negotiation scenarios.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28837">ASTERISK-28837</a>: pjproject_bundled: Honor --without-pjproject.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=966acc6251bc881f7118276d4dad063abfb24594">[966acc6251]</a> Alexander Traud -- pjproject_bundled: Honor --without-pjproject.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28758">ASTERISK-28758</a>: pjsip startup errors when using "with-ssl" configure option<br/>Reported by: Patrick Wakano<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3431949a5294bf6aaaddb640619d832f4d678a21">[3431949a52]</a> Alexander Traud -- pjproject_bundled: Repair ./configure --with-ssl without ARG.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26955">ASTERISK-26955</a>: pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected<br/>Reported by: Peter Sokolov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d9bde76a98b19e93b5e190fe5795c24b4195644">[9d9bde76a9]</a> Sean Bright -- pjproject_bundled: Allow brackets in via parameters</li>
-</ul><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=5d9f9f4871a6721b5f60132373ee68a52cd467df">[5d9f9f4871]</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=0844d6b127730460375cfa922046b42a1adff95f">[0844d6b127]</a> Dan Cropp -- pjproject: Configurable setting for cnonce to include hyphens or not</li>
-</ul><br><h3>Improvement</h3><h4>Category: Addons/chan_mobile</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28832">ASTERISK-28832</a>: chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio<br/>Reported by: Peter Turczak<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3303defd3fa01b2698a6aa766b43deb01aee3efa">[3303defd3f]</a> Peter Turczak -- chan_mobile: Add smoother to make SIP/RTP endpoints happy.</li>
-</ul><br><h4>Category: Applications/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28484">ASTERISK-28484</a>: Add AudioSocket support<br/>Reported by: Seán C. McCord<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=163efbd724e9691f4687ddb6bb6ecd6f7d028815">[163efbd724]</a> Seán C McCord -- feat: AudioSocket channel, application, and ARI support.</li>
-</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28658">ASTERISK-28658</a>: app_confbridge: Add support for setting maximum sample rate<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b7144fbdb1b892fca03d92fb56f9ff1843e443">[89b7144fbd]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
-</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24798">ASTERISK-24798</a>: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor<br/>Reported by: xrobau<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddfb60ac2c92e0f182151792a489059afde6bf57">[ddfb60ac2c]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
-</ul><br><h4>Category: Applications/app_page</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27946">ASTERISK-27946</a>: dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't<br/>Reported by: Joshua Elson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbddb6725d84071b43a92bd2036670a9ef8e461b">[dbddb6725d]</a> sungtae kim -- dial.c: Removed dial string 80 character limitation</li>
-</ul><br><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=7362647e2f29334cd1bb4236f2403222794124a8">[7362647e2f]</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=2d67dbfef5dcdff9825149968a66457fa9389b1a">[2d67dbfef5]</a> cmaj -- app_voicemail.c: Support multiple file formats for forwarded messages.</li>
-</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a5be92b792a2ba81cb8258947f28371bd767bf5">[5a5be92b79]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a5be92b792a2ba81cb8258947f28371bd767bf5">[5a5be92b79]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a5be92b792a2ba81cb8258947f28371bd767bf5">[5a5be92b79]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28658">ASTERISK-28658</a>: app_confbridge: Add support for setting maximum sample rate<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b7144fbdb1b892fca03d92fb56f9ff1843e443">[89b7144fbd]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
-</ul><br><h4>Category: Channels/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28484">ASTERISK-28484</a>: Add AudioSocket support<br/>Reported by: Seán C. McCord<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=163efbd724e9691f4687ddb6bb6ecd6f7d028815">[163efbd724]</a> Seán C McCord -- feat: AudioSocket channel, application, and ARI support.</li>
-</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28638">ASTERISK-28638</a>: Simplify dialplan for Dial, Page, and ChanIsAvail<br/>Reported by: cmaj<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe3cce816c3bc390a47ac25c868e357fccc96443">[fe3cce816c]</a> Richard Mudgett -- app_chanisavail.c: Simplify dialplan using ChanIsAvail.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=abcb4ab3216cb11f1142b9e890945a6a861148f9">[abcb4ab321]</a> Richard Mudgett -- app_dial.c: Simplify dialplan using Dial.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d86a6ac5ce11ae3df052672fe265cb828f7da327">[d86a6ac5ce]</a> Richard Mudgett -- app_page.c: Simplify dialplan using Page.</li>
-</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28726">ASTERISK-28726</a>: install_prereq script uses the interactive mode when installing aptitude<br/>Reported by: Sylvain Afchain<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c02d0a4500cbd4f657c26eb609928c110ebccb9">[0c02d0a450]</a> Sylvain Afchain -- install_prereq: Install aptitude non-interactively</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=c18983207d08b837b44cb1e82f4e5c3f8e84bac4">[c18983207d]</a> Florian Floimair -- core: Add H.265/HEVC passthrough support</li>
-</ul><br><h4>Category: Core/DNS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28853">ASTERISK-28853</a>: Missing include on FreeBSD<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97494d898473a416b99b4b54faaf612dc739e0dc">[97494d8984]</a> Guido Falsi -- core/dns: Add system include required on FreeBSD</li>
-</ul><br><h4>Category: Core/Dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27946">ASTERISK-27946</a>: dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't<br/>Reported by: Joshua Elson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbddb6725d84071b43a92bd2036670a9ef8e461b">[dbddb6725d]</a> sungtae kim -- dial.c: Removed dial string 80 character limitation</li>
-</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28750">ASTERISK-28750</a>: TLS/SSL Key too small error<br/>Reported by: Martin Zeh<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f2d56fc8c0068bdd172a558f9eebf0e81693c48">[7f2d56fc8c]</a> Sean Bright -- tcptls.c: Log more informative OpenSSL errors</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28710">ASTERISK-28710</a>: Should be able to disable the /httpstatus URI in the built-in HTTP server<br/>Reported by: Sean Bright<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dce6f746bed6340e5e11ecbb5bb9f2f7926537f">[0dce6f746b]</a> Sean Bright -- http: Add ability to disable /httpstatus URI</li>
-</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28945">ASTERISK-28945</a>: AMI SendText - add Content-Type parameter<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cfed0ea0333287d2b8503fd1a32d3622f235f533">[cfed0ea033]</a> Kevin Harwell -- manager - Add Content-Type parameter to the SendText action</li>
-</ul><br><h4>Category: Core/Streams</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a5be92b792a2ba81cb8258947f28371bd767bf5">[5a5be92b79]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24798">ASTERISK-24798</a>: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor<br/>Reported by: xrobau<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddfb60ac2c92e0f182151792a489059afde6bf57">[ddfb60ac2c]</a> Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28673">ASTERISK-28673</a>: GET FULL VARIABLE documentation clarification<br/>Reported by: Jonathan Harris<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d94bdde9d48b385a41c794a7260fd3773cf6ca5">[7d94bdde9d]</a> Sean Bright -- res_agi: Improve GET FULL VARIABLE documentation</li>
-</ul><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=0dc7e29dd84e636d9e03b3c0e14178b7fb843d6b">[0dc7e29dd8]</a> Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling correetions.</li>
-</ul><br><h4>Category: Functions/func_volume</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28813">ASTERISK-28813</a>: func_volume: Allow decimal numbers as parameter to improve granularity<br/>Reported by: Jean Aunis - Prescom<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de66713fd51d72d2b37908741e0aab9e4b4dfc7c">[de66713fd5]</a> Jean Aunis -- func_volume: Accept decimal number as argument</li>
-</ul><br><h4>Category: Resources/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28484">ASTERISK-28484</a>: Add AudioSocket support<br/>Reported by: Seán C. McCord<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=163efbd724e9691f4687ddb6bb6ecd6f7d028815">[163efbd724]</a> Seán C McCord -- feat: AudioSocket channel, application, and ARI support.</li>
-</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28629">ASTERISK-28629</a>: [patch] Add an "inhibitCOLP" flag to the bridges REST API<br/>Reported by: Jean Aunis - Prescom<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=034ac357adeefad2cea9af579be6391ab17866c0">[034ac357ad]</a> Jean Aunis -- ARI: Ability to inhibit COLP frames when adding channels to a bridge</li>
-</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28896">ASTERISK-28896</a>: ari: Add support for specifying variables on channel create<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=15cbff9d54b4f57068b5753e412fe82b98a74cb1">[15cbff9d54]</a> Joshua C. Colp -- ari: Allow variables to be set on channel create.</li>
-</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28958">ASTERISK-28958</a>: Continue reading string when ping received by websocket<br/>Reported by: Nickolay V. Shmyrev<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7163efd934b75a39c766313ec2eec05c1439dbf0">[7163efd934]</a> Nickolay Shmyrev -- res_http_websocket.c: Continue reading after ping/pong</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28949">ASTERISK-28949</a>: res_http_websocket: Add masking to websocket client<br/>Reported by: Moises Silva<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9445dac43b68ea3adff9c52cd8722f0adb86c079">[9445dac43b]</a> Moises Silva -- res_http_websocket: Add payload masking to the websocket client</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28959">ASTERISK-28959</a>: res_pjsip: Added option for disable rport parameter set<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=81b5e4a73f321b0e1761ef6661f883e7d85070b4">[81b5e4a73f]</a> sungtae kim -- res_pjsip.c: Added disable_rport option for pjsip.conf</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28777">ASTERISK-28777</a>: Codec Negotiation: add outgoing_call_offer_prefs option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ee455958ee200a03afb9ee01b0034a8ccbe4f39">[2ee455958e]</a> George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28756">ASTERISK-28756</a>: Codec Negotiation: add incoming_call_offer_pref option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06dada3f01c8fd43ad30dff0feb11cb248ff426d">[06dada3f01]</a> Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option</li>
-</ul><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=b43cdc7f1e14f350c4b4d15394ffc8e54460617c">[b43cdc7f1e]</a> Torrey Searle -- channel/chan_pjsip: add dialplan function for music on hold</li>
-</ul><br><h4>Category: Resources/res_pjsip_logger</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28895">ASTERISK-28895</a>: res_pjsip_logger: Add tons'o'functionality<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a7aaee70c65f52c6b65bd56c0b99ac72e1658e26">[a7aaee70c6]</a> Joshua C. Colp -- res_pjsip_logger: Expand functionality to improve logging.</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=e73eba85c16a40a1212ccb9f435617d68004a8da">[e73eba85c1]</a> Joshua Colp -- res_pjsip_outbound_registration: Extend documentation for "max_retries".</li>
-</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28777">ASTERISK-28777</a>: Codec Negotiation: add outgoing_call_offer_prefs option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ee455958ee200a03afb9ee01b0034a8ccbe4f39">[2ee455958e]</a> George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28756">ASTERISK-28756</a>: Codec Negotiation: add incoming_call_offer_pref option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06dada3f01c8fd43ad30dff0feb11cb248ff426d">[06dada3f01]</a> Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28733">ASTERISK-28733</a>: stream: Add support for adding/removing streams during SFU/calls<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a5be92b792a2ba81cb8258947f28371bd767bf5">[5a5be92b79]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28777">ASTERISK-28777</a>: Codec Negotiation: add outgoing_call_offer_prefs option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ee455958ee200a03afb9ee01b0034a8ccbe4f39">[2ee455958e]</a> George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28782">ASTERISK-28782</a>: Add support for Content-Disposition header in multi-part INVITES<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e12244153a245d19d7ebd32cb17208104c64f443">[e12244153a]</a> Torrey Searle -- res_pjsip_session: implement processing of Content-Disposition</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28787">ASTERISK-28787</a>: res_pjsip_session: Decide more intelligently when to add video<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21e905146105372e424ffcee35af983ed217cf5f">[21e9051461]</a> Joshua C. Colp -- res_pjsip_session: Apply intention behind requested formats.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28756">ASTERISK-28756</a>: Codec Negotiation: add incoming_call_offer_pref option<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06dada3f01c8fd43ad30dff0feb11cb248ff426d">[06dada3f01]</a> Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option</li>
-</ul><br><h4>Category: Third-Party/pjproject</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28866">ASTERISK-28866</a>: third-party/pjproject/configure.m4 contains bashisms<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c831f032730fbe1aac98b493b4d768d5195f8f60">[c831f03273]</a> Guido Falsi -- pjproject: Remove bashism from configure.m4 script</li>
-</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28899">ASTERISK-28899</a>: Upgrade Asterisk to bundled pjproject 2.10<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=415b55af5ab1a126f6d46f65808a00bb0306ba39">[415b55af5a]</a> Kevin Harwell -- pjproject: Upgrade bundled version to pjproject 2.10</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28879">ASTERISK-28879</a>: pjproject has race conditions in it's build system<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=801d570f6ec55e565e6b6291b4817a95430965d2">[801d570f6e]</a> Guido Falsi -- pjproject: Fix race condition when building with parallel make</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: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28798">ASTERISK-28798</a>: [patch] chan_sip: TCP/TLS client without server.<br/>Reported by: Alexander Traud<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da9554d925687b486d1fb623d9cfd40d5ed184a9">[da9554d925]</a> Alexander Traud -- chan_sip: TCP/TLS client without server.</li>
-</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28719">ASTERISK-28719</a>: Cannot remove defaultrule from queue using realtime queues<br/>Reported by: EDV O-TON<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb9252ea27098f18233f71797e877e956ab817f4">[eb9252ea27]</a> Sean Bright -- res_config_odbc: Preserve empty strings returned by the database</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=6fd94258f808a92243f8a87a63f4a4e2e86b80f2">6fd94258f8</a></td><td>Asterisk Development Team</td><td>Update for 18.0.0-rc2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=704cb887992dcdbb930e79144d093d5efc24d568">704cb88799</a></td><td>Asterisk Development Team</td><td>Update for 18.0.0-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f589985840b95004aa5188a2d6a1b6920520648a">f589985840</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.0.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a49757e400fa73985f84f3290c769dd2e35c304">5a49757e40</a></td><td>Patrick Verzele</td><td>res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec03909831364712c125e6bde51364ece685fedf">ec03909831</a></td><td>Kevin Harwell</td><td>conversions: Add string to signed integer conversion functions</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e32815dddb3170fbe1f88567edd6b867d8681690">e32815dddb</a></td><td>George Joseph</td><td>ast_coredumper: Fix issues with naming</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ed1b1452daab54a0eab0d3a66e95d31609b2952">9ed1b1452d</a></td><td>Alexander Traud</td><td>sip_nat_settings: Update script for latest Linux.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=217449a1e5ce522e787ceb99b52aa663f2d12edf">217449a1e5</a></td><td>Alexander Traud</td><td>samples: Fix keep_alive_interval default in pjsip.conf.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a8cacb93db983e97f5aaf429a514f8106af8c9e">5a8cacb93d</a></td><td>George Joseph</td><td>logger.c: Added a new log formatter called "plain"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5dfeeba623361ec4460b4791b5bab1b62ccc770b">5dfeeba623</a></td><td>Sean Bright</td><td>res_musiconhold.c: Use ast_file_read_dir to scan MoH directory</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c4c72d55a2d5d68226cc33fb985f7c9b15b6ca1f">c4c72d55a2</a></td><td>George Joseph</td><td>scope_trace: Added debug messages and added additional macros</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d26ab7f8f93f543d86047f55fe1d3cbe18a6285f">d26ab7f8f9</a></td><td>George Joseph</td><td>stream.c: Added 2 more debugging utils and added pos to stream string</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6faf76308d80ad3d32fc7379d21b9dc920579fb6">6faf76308d</a></td><td>George Joseph</td><td>ACN: Changes specific to the core</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3d87f78eda78e7e5482f312c252eec733248c0b">a3d87f78ed</a></td><td>Joshua C. Colp</td><td>res_pjsip: Fix codec preference defaults.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da8a617dc9863b06f1e74ce1e82e8b460860d7b1">da8a617dc9</a></td><td>Sean Bright</td><td>vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=769a9611e7639b02e12a5bbb1b7fdc72f96b251d">769a9611e7</a></td><td>Ben Ford</td><td>utils.c: NULL terminate ast_base64decode_string.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=802aa97fa044ac9fb6d32912ae7372e05dfe2bcf">802aa97fa0</a></td><td>George Joseph</td><td>ACN: Configuration renaming for pjsip endpoint</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de23cb4002273d7f2e8b6462c72f52cf2c13d171">de23cb4002</a></td><td>Ben Ford</td><td>res_stir_shaken: Fix memory allocation error in curl.c</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71446b68fc17f09253272b8d76a5118a553ced56">71446b68fc</a></td><td>George Joseph</td><td>res_pjsip_session: Ensure reused streams have correct bundle group</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d9ae902f524c8e8509d9fcb0a106a7a1be246447">d9ae902f52</a></td><td>Sean Bright</td><td>utf8.c: Add UTF-8 validation and utility functions</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9022f35f098ae5ba87efab9dba66723512c786df">9022f35f09</a></td><td>Sean Bright</td><td>vector.h: Add AST_VECTOR_SORT()</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a678dafac8f6fe5403a86742b04db6b8a0a118fe">a678dafac8</a></td><td>George Joseph</td><td>CI: Force publishAsteriskDocs to use python2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af70bbb13af78242027c7a3eda04c6abc2fbc72b">af70bbb13a</a></td><td>Joshua C. Colp</td><td>websocket / pjsip: Increase maximum packet size.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d15f7272102f39d0f64093f6f30bc62ac28e382">8d15f72721</a></td><td>Joshua C. Colp</td><td>pjsip: Include timer patch to prevent cancelling timer 0.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=333076421337b7f570bdb727c2ecdbb0e243b187">3330764213</a></td><td>George Joseph</td><td>Update .gitreview defaultbranch to 18</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f5e6805bfb6c32371e53b375412514afc7a5b63">1f5e6805bf</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 18.0.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5fbed5af24b8296b47cb0d6371f8187c1ee205cb">5fbed5af24</a></td><td>Ben Ford</td><td>res_stir_shaken: Add stir_shaken option and general improvements.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e88beedd0845aa18a2f5594c6638d443399c93a9">e88beedd08</a></td><td>George Joseph</td><td>res_pjsip_session: Fix segv in session_on_rx_response</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9bd1d686a120db5233acb2d6f90c31d4f52464a0">9bd1d686a1</a></td><td>George Joseph</td><td>ACN: Add tracing to existing code</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d22e342060d4541576f82f47d80b7a1f88c29e8">2d22e34206</a></td><td>George Joseph</td><td>ACN: res_pjsip endpoint options</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d093e44b1ebe77e553d112b34fa0c1b141802405">d093e44b1e</a></td><td>George Joseph</td><td>frame.c: Make debugging easier</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=955b7b4fdbed9ca2d79091c3d33b8331c1fdf7ff">955b7b4fdb</a></td><td>George Joseph</td><td>Scope Trace: Make it easier to trace through synchronous tasks</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d1064eaafe899044adbd30a40cdaa15c315c463">8d1064eaaf</a></td><td>George Joseph</td><td>Streams: Add features for Advanced Codec Negotiation</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7440fd03970d831a5db80c740349b30efe5f283e">7440fd0397</a></td><td>George Joseph</td><td>Scope Trace: Add some new tracing macros and an ast_str helper</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=12741171022220a24786556eaf97a0b90ddf7b24">1274117102</a></td><td>Ben Ford</td><td>res_stir_shaken: Add outbound INVITE support.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1cfd549767b93ae3c3789bbfad6009f14e11ab1">f1cfd54976</a></td><td>Walter Doekes</td><td>res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9f42a717efec91229d22e04fd9f0e356a58e1c0">b9f42a717e</a></td><td>George Joseph</td><td>app_confbridge: Plug ref leak of bridge channel with send_events</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d1bf3c537bba0416f691f48165fdd0a32554e8a">3d1bf3c537</a></td><td>Kevin Harwell</td><td>Compiler fixes for gcc 10</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=559fa0e89c09efcd90380949e0975091914634ad">559fa0e89c</a></td><td>Ben Ford</td><td>cli.c: Fix compiler error.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3927f79cb51d83c5880ce1d86d7475f3cf0fbc89">3927f79cb5</a></td><td>Ben Ford</td><td>res_stir_shaken: Add inbound INVITE support.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1fcb6b1b21484473b10f1bc02dc9f16bb3e76683">1fcb6b1b21</a></td><td>Joshua C. Colp</td><td>bridge_channel: Don't queue unmapped frames.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ca3c22c5f1fd042a3d42a2f421d37541e4efd47a">ca3c22c5f1</a></td><td>George Joseph</td><td>Scope Tracing: A new facility for tracing scope enter/exit</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec7890d7c63105368e5d377b518bee77abdc45a8">ec7890d7c6</a></td><td>Joshua C. Colp</td><td>res_sorcery_config: Always reload configuration on errors.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f506cc4896d1cf6428c376a98470ede0760e86f1">f506cc4896</a></td><td>Ben Ford</td><td>res_stir_shaken: Add unit tests for signing and verification.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e29df34de03d66f1371e1ba937e2b2155e069844">e29df34de0</a></td><td>Ben Ford</td><td>res_stir_shaken: Added dialplan function and API call.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=44e5dd288b5ecf2e7a1c1e99f87da5e00485f67f">44e5dd288b</a></td><td>Jaco Kroon</td><td>Remove #include <sys/cdefs.h></td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1cfd30bd8a998a9c3d3827944aa6c39fdf4db7e8">1cfd30bd8a</a></td><td>Joshua C. Colp</td><td>res_stir_shaken: Use ast_asprintf for creating file path.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9acf840f7c5d8393777b844b15b1114d1d55c47c">9acf840f7c</a></td><td>Ben Ford</td><td>res_stir_shaken: Implemented signature verification.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7baf2c4bf1347f7ba8bf0fc9415ae08fe4b0ed5b">7baf2c4bf1</a></td><td>George Joseph</td><td>app_voicemail: Add workaround for a gcc 10 issue with -Wrestrict</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ef5ba58f545937af5d7af010c2206c7f25001b1">4ef5ba58f5</a></td><td>Alexander Traud</td><td>BuildSystem: Only if found LibPRI, check its optional parts.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ef580f96e71be055d44dbe07ff63fe39ec697408">ef580f96e7</a></td><td>Alexander Traud</td><td>BuildSystem: Only if found external PJProject, check its optional parts.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=611529fa527c62877162eb27e034ce646e7c925b">611529fa52</a></td><td>Alexander Traud</td><td>res_stir_shaken: Do not build without OpenSSL.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27de0c9700fdb9a54c7e39e220114313088c96d2">27de0c9700</a></td><td>Alexander Traud</td><td>res_audiosocket: Avoid Sometimes-uninitialized Warning with Clang.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b80e5f5daa029217c8b4ec382f62d8c19045800">2b80e5f5da</a></td><td>Jaco Kroon</td><td>res_rtp_asterisk: iterate all local addresses looking to populate ICE.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1cf569ba2b9155ca029729974d0b436db5bd26cf">1cf569ba2b</a></td><td>Jaco Kroon</td><td>res_pjsip: document legal dtls_verify endpoint options.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52ecbbd014002c0e040b9eeeb9bf2ceffba20903">52ecbbd014</a></td><td>Alexander Traud</td><td>_pjsua: Build even with Clang.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ee1c7f465b4c56a81fa944839bb1035310b330ff">ee1c7f465b</a></td><td>Alexander Traud</td><td>res_rtp_asterisk: Build without PJProject.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60925c68e8831f902ee99876e61997ecca9c3d0f">60925c68e8</a></td><td>Sean Bright</td><td>Revert "res_config_odbc: Preserve empty strings returned by the database"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c5f3836bcc532f4347601bb88a0916ef2011dd79">c5f3836bcc</a></td><td>Jaco Kroon</td><td>main/backtrace: binutils-2.34 fix.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ba6d43083d527406c99900bf286a01eb32d19b2">7ba6d43083</a></td><td>George Joseph</td><td>test_res_pjsip_session_caps: Create unit test</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=57a457c26ca00edbd44da71efa0fd20c26c8d293">57a457c26c</a></td><td>Ben Ford</td><td>res_stir_shaken: Implemented signing of JSON payload.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d32e559e8a04fbe403db5f69358e56d512ae26e9">d32e559e8a</a></td><td>Jaco Kroon</td><td>acl: implement a centralized ACL output mechanism for HAs and ACLs.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b6c58896f88707ef514eed0a077b3f7061cf3e6">1b6c58896f</a></td><td>Joshua C. Colp</td><td>chan_sip: Send 403 when ACL fails.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3ed80fc57b1e175aa3a47f0353a693132cdce7d8">3ed80fc57b</a></td><td>Joshua C. Colp</td><td>CHANGES: Change md file extension to txt.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26713dc88ba5c01d620f9ac2d2158fcbd2bb8cad">26713dc88b</a></td><td>Kevin Harwell</td><td>ast_coredumper: add Asterisk information dump</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6f731f153baec59562b0a8252b8a2f0fe668f6eb">6f731f153b</a></td><td>Jaco Kroon</td><td>netsock2: compile fixes.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=211bb8a79c66895ad1fd8cad60216f5560638f30">211bb8a79c</a></td><td>Ben Ford</td><td>res_stir_shaken: Initial commit and reading private key.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a699e016ddf730ba8296acc705c64e30b619c52e">a699e016dd</a></td><td>Jaco Kroon</td><td>build: enable building with uClibc</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f824cd6a13ee69a1c9d56380557e0ade56141d41">f824cd6a13</a></td><td>Jaco Kroon</td><td>build: search from newest to oldest for gmime.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82c3939c3825f411ff7ffc9e0404b9c1db6a0dc8">82c3939c38</a></td><td>Jaco Kroon</td><td>res_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ad64e97c05f613fc2dc11032bed41328d84d05e">2ad64e97c0</a></td><td>Jaco Kroon</td><td>Update main/backtrace.c to deal with changes in binutils 2.34.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49cf84578e630db55cd6cb368cc29ecc6df5f50e">49cf84578e</a></td><td>Sean Bright</td><td>chan_vpb: Fix 'catching polymorphic type ... by value' error</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d68f940f6e4b35b6d49cd3af048b3c828799bfd4">d68f940f6e</a></td><td>Sean Bright</td><td>dns_txt: Add TXT record parsing support</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=00a7e4b51df0202ab4168963b16ee51a5b69fb15">00a7e4b51d</a></td><td>George Joseph</td><td>CI: Create generic jenkinsfile</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e089779908a08f29b16e8b70c080fd2338b92d3b">e089779908</a></td><td>Rodrigo Ramírez Norambuena</td><td>res_rtp_asterisk: Add 'rtp show settings' cli command</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=680e6b9774d76ed264545e90f3bc0117e3ce321e">680e6b9774</a></td><td>Walter Doekes</td><td>app_queue: Refactor odd placement of if's around say_position</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e1651b4f44b5ae00136d1f393643e3cd39842c9">1e1651b4f4</a></td><td>Kevin Harwell</td><td>format_cap: make function parameters 'const'</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b5c6fddf16afab149a0aadef8f0130a896bf720">0b5c6fddf1</a></td><td>Walter Doekes</td><td>say: Remove unused "plural" option from main/say</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5cd7230f3c3db1cc3429612580c7a56c113464ad">5cd7230f3c</a></td><td>Jaco Kroon</td><td>addons/res_config_mysql: silense warnings about printf format errors.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de6919f33942911647b1ec0eccfdd942ad776f55">de6919f339</a></td><td>Sean Bright</td><td>ast_tls_cert: Allow private key size to be set on command line</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8dcdce42a90724d9a50ba06a973fb778db583deb">8dcdce42a9</a></td><td>Sean Bright</td><td>app_mixmonitor: Turn on synchronization by default</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f6ee98c3f70fbfa20fc34df132fee566e8188a9">0f6ee98c3f</a></td><td>Joshua C. Colp</td><td>stasis: Use format specifier for size_t.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e037ebb973c344537e24ab5cc31b4ead42c3905">1e037ebb97</a></td><td>Sean Bright</td><td>func_odbc: Prevent snprintf() truncation warning</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a72caa041f10ed870fcb3038a561be5295157e3d">a72caa041f</a></td><td>George Joseph</td><td>doc: Fix CHANGES entries to have .txt suffix and update READMEs</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b53d329ac648ec958e8c600b5b47787643dd30c">1b53d329ac</a></td><td>Joshua C. Colp</td><td>res_rtp_asterisk: Don't produce transport-cc if no packets.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b76ab5e5c9dc51f88a5ee5c101410124b8283cfe">b76ab5e5c9</a></td><td>George Joseph</td><td>message.c: Add option to suppress the Message channel AMI and ARI events</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=113d05e504ea72db20fc09766f1c7e79d2469533">113d05e504</a></td><td>Walter Doekes</td><td>chan_sip: Clarify in sample docs how directmediapermit/-acl should be used</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=262221f4d9973a8a175ca8f8ae3de91add284bde">262221f4d9</a></td><td>Sean Bright</td><td>func_odbc.conf.sample: Add example lookup</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f09cf4da4479221815357f021ad0f9f4d7acf461">f09cf4da44</a></td><td>Sean Bright</td><td>app_voicemail: Remove MessageExists and MESSAGE_EXISTS()</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5cbf47714a8dc35f6e412e3b2e53deee735f3aef">5cbf47714a</a></td><td>Sean Bright</td><td>app_voicemail, say: Fix various leading whitespace problems</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3bc8b36537046bebedccc2cb683ec16c64241485">3bc8b36537</a></td><td>Jaco Kroon</td><td>netsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=00a7432156aec3239017a39cf459af34e427d78d">00a7432156</a></td><td>Kevin Harwell</td><td>app_agent_pool: Update XML docs for AgentLogin</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=19069f7db75a00fd6a2529009f67f9a3d9c7bb2c">19069f7db7</a></td><td>Richard Mudgett</td><td>app_bridgeaddchan.c: Make BridgeAdd be more like Bridge</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0376f2bba94770d480428e9d944a9a3a6139cb52">0376f2bba9</a></td><td>Richard Mudgett</td><td>features.c: Make Bridge application tolerate unspecified channel.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d1f3d9bf3118f0bb6c5715325afa83a0c07f352">0d1f3d9bf3</a></td><td>Richard Mudgett</td><td>app_chanspy.c: Reduce log message level from notice to verbose.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a45794719871496249386744a471e8f6f8d8ddba">a457947198</a></td><td>Richard Mudgett</td><td>app_softhangup.c: Reduce unnecessary warning to verbose message.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc99ac8c9aaa9e0119f237909236a35c805e5732">fc99ac8c9a</a></td><td>Sean Bright</td><td>db: Initialize condition primitive before use</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32160cb45691b11b8a059688bee65a7146bde29e">32160cb456</a></td><td>Jaco Kroon</td><td>ACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d0b198b330e20de9652c0b5b461733ea461544ff">d0b198b330</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=0183e2bc67e9feaed5a6798ed7631346ac8a95da">0183e2bc67</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=fd823225a6e80cc3974c9ca45eb74d433a7d689b">fd823225a6</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=366da90f74cf2d6010e031650d013ff3f97fa7b4">366da90f74</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=91c3b5b09d0b05355ed8245cf645c86dfcccab7f">91c3b5b09d</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=48161dfc71a66154444392b70243db55de8bbc78">48161dfc71</a></td><td>Rodrigo Ramírez Norambuena</td><td>queue_log: Add alembic script for generate db table for queue_log</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a6a2800e797116d4c50901f31dae90284308d6e">2a6a2800e7</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=4abb54b2e4048a252b67375cb1c6bd6fd6f75769">4abb54b2e4</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=e8e1314fcb42d7c59f1db149fdca1ebf04492b94">e8e1314fcb</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=a5fa0d662e1f56f8c83efcca18218bf3dd7e9cc9">a5fa0d662e</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=f2d5ed54eaa8158fe49aeb5a7f2ecad986018c0e">f2d5ed54ea</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=bdd785d31cc1abfdbe33c1a8fc7ca67c04293687">bdd785d31c</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=0e3b397812e12047b7956559af6ff4cda6039ec7">0e3b397812</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=8a1f30af0451c22989e1a4e832eeaf8b337f8a89">8a1f30af04</a></td><td>Corey Farrell</td><td>core: Improve MALLOC_DEBUG for frames.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d71d0f9489b29450a202193ab009d5539f01541c">d71d0f9489</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=6e907ae5d440c0003860f998592a919330657bbe">6e907ae5d4</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=a60d2e905c4c263a2e05965b92c7f46e4e260587">a60d2e905c</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=b27a5183da84f59629f47961437cf8c6bf8a91fe">b27a5183da</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=c0efe19cec7234df8785cc6efd672e9ca15179e3">c0efe19cec</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=2970a13fb8f03352af4045074a20b99e8322b9e7">2970a13fb8</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=068ed2c626de575d65d78f11e9d7f1b68014bb3a">068ed2c626</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=ba64d68273ecd2425514658e5e02c059506d16aa">ba64d68273</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=4c3655ecfdd7b23dad6f2a909cad06dc8e0b38f7">4c3655ecfd</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=966488ab52e16f36ca86c72e3c8b355ffebda74f">966488ab52</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=f7045cefd900f64988bca9ae3233be0433bda9a4">f7045cefd9</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=67ba62f4e6d4bc32ee503839889f5a47932fb9d8">67ba62f4e6</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=4de1e6d0e6f85e715b83dc9772fec871c9c0efb6">4de1e6d0e6</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=725e991fafb7ef1cc884741e552019c1881b4b63">725e991faf</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=e82f2f6e82c05016125b67e5fb1e5b4e0885b2dd">e82f2f6e82</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=a4142c84373003e8f9aece4653aa8468b3f2882a">a4142c8437</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=ca608d25756142be53af963cba0067b2bdc17f52">ca608d2575</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=3dfbc05c534e498e2196a0f6ecd8c64a9da1115d">3dfbc05c53</a></td><td>Corey Farrell</td><td>stasis: Fix leaks</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=863fe2225f3954dc1aeacf8fce44cd7e61756524">863fe2225f</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=723b695ce50684aac8a707edaa0f749c530ffe2f">723b695ce5</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=0e56643d9fa2356fa4114a18eda1f3a49f5b96b1">0e56643d9f</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=2ae1a22e0e71fc705ffd93d4ac061c48b0bb593c">2ae1a22e0e</a></td><td>George Joseph</td><td>ARI: External Media</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5fb9b23105815f03973d3da4b22e30580ca95116">5fb9b23105</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=ed757cc7bbc1daa90ee692923f9fdf324e84ac8e">ed757cc7bb</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=1d06a1efb3819ba2a9cf933113523a73fc1ce021">1d06a1efb3</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=649003821eefd4f89f3b8b55f3576ab8c7823180">649003821e</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=3ef52b0b176715283a2520d523199f3c19811385">3ef52b0b17</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=19045db39296dfb2ed2dc42814b0199f0bc45b54">19045db392</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=9e015713cc7e17428528db80716923658a22e62c">9e015713cc</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=8da4e28a81e46eb1e5be0702849e1304fce2764f">8da4e28a81</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=64906c4c9ba63e18f2c71310fdbf14450dac7b62">64906c4c9b</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=446bac733d5f2bb02ded1c34a3301a08b8c9255e">446bac733d</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=be6130607dc982ddd9a6a813f61a93db45840e0a">be6130607d</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=c01dd2a41a39df3f8d46ea8240d799618fe0276f">c01dd2a41a</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=9d07d5a6d6157126137c736200d63b9756c2a53f">9d07d5a6d6</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=1f8ae708a030afafa687923261e28c2eb5c7d425">1f8ae708a0</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=5f66fb5139c2dcaf1c47ebebac20b79972e315fd">5f66fb5139</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=5e6e1175d5f09a39fdc5b9af03ebeab08ddbaad4">5e6e1175d5</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=8d10028b98905fdb8ef3015ab56d8a0060602a44">8d10028b98</a></td><td>George Joseph</td><td>Update master for Asterisk 18</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ce9ee7f2e6e31695516268dcb62bebe0e1264a3">7ce9ee7f2e</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=8e44d823c17255c9714b6c72ce804474ee23c58c">8e44d823c1</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>asterisk-17.0.0-summary.html | 1265 --
-asterisk-17.0.0-summary.txt | 2973 ----
-b/.gitreview | 2
-b/.version | 2
-b/CHANGES | 400
-b/ChangeLog | 6317 +++++++++-
-b/Makefile | 8
-b/Makefile.rules | 19
-b/README-SERIOUSLY.bestpractices.md | 4
-b/UPGRADE.txt | 120
-b/addons/cdr_mysql.c | 18
-b/addons/chan_mobile.c | 63
-b/addons/chan_ooh323.c | 7
-b/addons/ooh323c/src/decode.c | 10
-b/addons/ooh323c/src/ooSocket.c | 2
-b/addons/ooh323c/src/oochannels.c | 2
-b/addons/res_config_mysql.c | 16
-b/apps/app_agent_pool.c | 4
-b/apps/app_amd.c | 24
-b/apps/app_audiosocket.c | 240
-b/apps/app_bridgeaddchan.c | 62
-b/apps/app_cdr.c | 8
-b/apps/app_chanisavail.c | 137
-b/apps/app_chanspy.c | 3
-b/apps/app_confbridge.c | 9
-b/apps/app_dahdiras.c | 6
-b/apps/app_dial.c | 108
-b/apps/app_dictate.c | 4
-b/apps/app_fax.c | 2
-b/apps/app_followme.c | 10
-b/apps/app_getcpeid.c | 1
-b/apps/app_meetme.c | 2
-b/apps/app_minivm.c | 3
-b/apps/app_mixmonitor.c | 51
-b/apps/app_osplookup.c | 13
-b/apps/app_page.c | 30
-b/apps/app_playback.c | 9
-b/apps/app_queue.c | 201
-b/apps/app_readexten.c | 3
-b/apps/app_record.c | 3
-b/apps/app_senddtmf.c | 13
-b/apps/app_softhangup.c | 2
-b/apps/app_stack.c | 11
-b/apps/app_stream_echo.c | 2
-b/apps/app_voicemail.c | 839 -
-b/apps/confbridge/conf_config_parser.c | 30
-b/apps/confbridge/confbridge_manager.c | 2
-b/apps/confbridge/include/confbridge.h | 2
-b/asterisk-18.0.0-rc2-summary.html | 15
-b/asterisk-18.0.0-rc2-summary.txt | 83
-b/bridges/bridge_native_rtp.c | 173
-b/bridges/bridge_simple.c | 200
-b/bridges/bridge_softmix.c | 281
-b/cdr/cdr_odbc.c | 2
-b/cdr/cdr_pgsql.c | 2
-b/cel/cel_pgsql.c | 2
-b/channels/Makefile | 2
-b/channels/chan_audiosocket.c | 302
-b/channels/chan_dahdi.c | 39
-b/channels/chan_dahdi.h | 18
-b/channels/chan_iax2.c | 16
-b/channels/chan_motif.c | 9
-b/channels/chan_pjsip.c | 314
-b/channels/chan_rtp.c | 19
-b/channels/chan_sip.c | 268
-b/channels/chan_unistim.c | 178
-b/channels/chan_vpb.cc | 2
-b/channels/iax2/parser.c | 18
-b/channels/pjsip/cli_commands.c | 13
-b/channels/pjsip/dialplan_functions.c | 84
-b/channels/pjsip/include/dialplan_functions.h | 25
-b/channels/sig_pri.c | 25
-b/channels/sip/include/sip.h | 1
-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/asterisk.conf.sample | 6
-b/configs/samples/confbridge.conf.sample | 7
-b/configs/samples/extconfig.conf.sample | 1
-b/configs/samples/func_odbc.conf.sample | 19
-b/configs/samples/http.conf.sample | 10
-b/configs/samples/indications.conf.sample | 11
-b/configs/samples/logger.conf.sample | 15
-b/configs/samples/musiconhold.conf.sample | 23
-b/configs/samples/pjsip.conf.sample | 79
-b/configs/samples/queues.conf.sample | 2
-b/configs/samples/rtp.conf.sample | 30
-b/configs/samples/sip.conf.sample | 4
-b/configs/samples/stir_shaken.conf.sample | 61
-b/configure | 666 -
-b/configure.ac | 196
-b/contrib/ast-db-manage/README.md | 1
-b/contrib/ast-db-manage/config/versions/61797b9fced6_add_stir_shaken.py | 31
-b/contrib/ast-db-manage/config/versions/79290b511e4b_pjsip_add_disable_rport.py | 39
-b/contrib/ast-db-manage/config/versions/b80485ff4dd0_add_pjsip_endpoint_acn_options.py | 29
-b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py | 54
-b/contrib/ast-db-manage/queue_log.ini.sample | 58
-b/contrib/ast-db-manage/queue_log/env.py | 1
-b/contrib/ast-db-manage/queue_log/script.py.mako | 24
-b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py | 38
-b/contrib/realtime/mysql/mysql_config.sql | 39
-b/contrib/realtime/postgresql/postgresql_config.sql | 45
-b/contrib/scripts/ast_coredumper | 421
-b/contrib/scripts/ast_tls_cert | 8
-b/contrib/scripts/install_prereq | 30
-b/contrib/scripts/sip_nat_settings | 19
-b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
-b/contrib/valgrind.supp | 14
-b/doc/CHANGES-staging/README.md | 8
-b/doc/CHANGES-staging/hide_messaging_ami_events | 11
-b/doc/UPGRADE-staging/README.md | 7
-b/doc/appdocsxml.dtd | 2
-b/formats/format_g726.c | 16
-b/formats/msgsm.h | 4
-b/funcs/func_aes.c | 4
-b/funcs/func_channel.c | 8
-b/funcs/func_curl.c | 49
-b/funcs/func_env.c | 5
-b/funcs/func_jitterbuffer.c | 19
-b/funcs/func_odbc.c | 26
-b/funcs/func_pjsip_contact.c | 6
-b/funcs/func_volume.c | 12
-b/include/asterisk/abstract_jb.h | 4
-b/include/asterisk/acl.h | 69
-b/include/asterisk/app.h | 7
-b/include/asterisk/ari.h | 2
-b/include/asterisk/astobj2.h | 61
-b/include/asterisk/audiohook.h | 2
-b/include/asterisk/autoconfig.h.in | 9
-b/include/asterisk/bridge.h | 21
-b/include/asterisk/bridge_features.h | 4
-b/include/asterisk/calendar.h | 4
-b/include/asterisk/channel.h | 87
-b/include/asterisk/channel_internal.h | 5
-b/include/asterisk/config.h | 18
-b/include/asterisk/config_options.h | 2
-b/include/asterisk/conversions.h | 54
-b/include/asterisk/core_unreal.h | 19
-b/include/asterisk/dns_core.h | 22
-b/include/asterisk/dns_internal.h | 31
-b/include/asterisk/dns_txt.h | 64
-b/include/asterisk/format_cache.h | 5
-b/include/asterisk/format_cap.h | 15
-b/include/asterisk/frame.h | 40
-b/include/asterisk/http_websocket.h | 14
-b/include/asterisk/iostream.h | 14
-b/include/asterisk/logger.h | 426
-b/include/asterisk/manager.h | 2
-b/include/asterisk/max_forwards.h | 1
-b/include/asterisk/message.h | 31
-b/include/asterisk/mixmonitor.h | 5
-b/include/asterisk/netsock2.h | 42
-b/include/asterisk/options.h | 7
-b/include/asterisk/parking.h | 5
-b/include/asterisk/res_audiosocket.h | 87
-b/include/asterisk/res_fax.h | 3
-b/include/asterisk/res_pjsip.h | 106
-b/include/asterisk/res_pjsip_presence_xml.h | 5
-b/include/asterisk/res_pjsip_pubsub.h | 23
-b/include/asterisk/res_pjsip_session.h | 27
-b/include/asterisk/res_pjsip_session_caps.h | 82
-b/include/asterisk/res_stir_shaken.h | 117
-b/include/asterisk/rtp_engine.h | 111
-b/include/asterisk/say.h | 4
-b/include/asterisk/serializer.h | 85
-b/include/asterisk/slin.h | 5
-b/include/asterisk/sorcery.h | 27
-b/include/asterisk/stasis.h | 3
-b/include/asterisk/stasis_app.h | 10
-b/include/asterisk/stasis_channels.h | 8
-b/include/asterisk/stream.h | 463
-b/include/asterisk/strings.h | 53
-b/include/asterisk/taskprocessor.h | 9
-b/include/asterisk/utf8.h | 188
-b/include/asterisk/utils.h | 59
-b/include/asterisk/vector.h | 50
-b/main/Makefile | 1
-b/main/abstract_jb.c | 178
-b/main/acl.c | 105
-b/main/app.c | 26
-b/main/ast_expr2.c | 1
-b/main/ast_expr2.y | 1
-b/main/asterisk.c | 41
-b/main/asterisk.exports.in | 1
-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 | 20
-b/main/backtrace.c | 9
-b/main/bridge.c | 15
-b/main/bridge_channel.c | 62
-b/main/cdr.c | 15
-b/main/channel.c | 204
-b/main/channel_internal_api.c | 12
-b/main/cli.c | 261
-b/main/codec_builtin.c | 8
-b/main/config.c | 28
-b/main/conversions.c | 51
-b/main/core_local.c | 112
-b/main/core_unreal.c | 141
-b/main/data_buffer.c | 2
-b/main/db.c | 3
-b/main/dial.c | 14
-b/main/dns_core.c | 75
-b/main/dns_srv.c | 6
-b/main/dns_txt.c | 127
-b/main/enum.c | 104
-b/main/event.c | 17
-b/main/features.c | 38
-b/main/file.c | 46
-b/main/format_cache.c | 8
-b/main/format_cap.c | 24
-b/main/frame.c | 160
-b/main/http.c | 62
-b/main/indications.c | 6
-b/main/iostream.c | 14
-b/main/logger.c | 150
-b/main/manager.c | 210
-b/main/media_cache.c | 47
-b/main/message.c | 55
-b/main/named_acl.c | 9
-b/main/options.c | 11
-b/main/pbx.c | 35
-b/main/pbx_variables.c | 31
-b/main/rtp_engine.c | 137
-b/main/say.c | 968 -
-b/main/serializer.c | 189
-b/main/sorcery.c | 46
-b/main/stasis.c | 53
-b/main/stasis_cache.c | 10
-b/main/stasis_channels.c | 7
-b/main/stasis_state.c | 298
-b/main/strcompat.c | 94
-b/main/stream.c | 432
-b/main/taskprocessor.c | 219
-b/main/tcptls.c | 33
-b/main/translate.c | 8
-b/main/utf8.c | 380
-b/main/utils.c | 50
-b/makeopts.in | 6
-b/menuselect/configure | 14
-b/menuselect/menuselect.c | 14
-b/menuselect/menuselect.h | 2
-b/pbx/pbx_dundi.c | 21
-b/res/Makefile | 2
-b/res/ari/ari_model_validators.c | 59
-b/res/ari/ari_model_validators.h | 23
-b/res/ari/config.c | 10
-b/res/ari/resource_bridges.c | 1
-b/res/ari/resource_bridges.h | 2
-b/res/ari/resource_channels.c | 245
-b/res/ari/resource_channels.h | 46
-b/res/ari/resource_events.c | 10
-b/res/parking/parking_bridge.c | 36
-b/res/parking/parking_bridge_features.c | 14
-b/res/parking/res_parking.h | 5
-b/res/res_agi.c | 20
-b/res/res_ari_bridges.c | 7
-b/res/res_ari_channels.c | 147
-b/res/res_ari_events.c | 2
-b/res/res_audiosocket.c | 345
-b/res/res_audiosocket.exports.in | 4
-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 | 7
-b/res/res_config_pgsql.c | 2
-b/res/res_corosync.c | 564
-b/res/res_fax.c | 37
-b/res/res_http_websocket.c | 97
-b/res/res_musiconhold.c | 472
-b/res/res_phoneprov.c | 6
-b/res/res_pjsip.c | 492
-b/res/res_pjsip/config_system.c | 8
-b/res/res_pjsip/config_transport.c | 17
-b/res/res_pjsip/location.c | 25
-b/res/res_pjsip/pjsip_configuration.c | 209
-b/res/res_pjsip/pjsip_message_filter.c | 40
-b/res/res_pjsip/pjsip_options.c | 7
-b/res/res_pjsip/pjsip_resolver.c | 30
-b/res/res_pjsip_acl.c | 20
-b/res/res_pjsip_caller_id.c | 3
-b/res/res_pjsip_config_wizard.c | 7
-b/res/res_pjsip_dialog_info_body_generator.c | 80
-b/res/res_pjsip_diversion.c | 7
-b/res/res_pjsip_dlg_options.c | 3
-b/res/res_pjsip_dtmf_info.c | 1
-b/res/res_pjsip_empty_info.c | 1
-b/res/res_pjsip_endpoint_identifier_ip.c | 102
-b/res/res_pjsip_history.c | 4
-b/res/res_pjsip_logger.c | 451
-b/res/res_pjsip_messaging.c | 58
-b/res/res_pjsip_mwi.c | 339
-b/res/res_pjsip_nat.c | 110
-b/res/res_pjsip_notify.c | 22
-b/res/res_pjsip_one_touch_record_info.c | 1
-b/res/res_pjsip_outbound_registration.c | 66
-b/res/res_pjsip_path.c | 1
-b/res/res_pjsip_pubsub.c | 148
-b/res/res_pjsip_refer.c | 8
-b/res/res_pjsip_registrar.c | 80
-b/res/res_pjsip_rfc3326.c | 1
-b/res/res_pjsip_sdp_rtp.c | 283
-b/res/res_pjsip_session.c | 667 -
-b/res/res_pjsip_session.exports.in | 1
-b/res/res_pjsip_session/pjsip_session_caps.c | 164
-b/res/res_pjsip_stir_shaken.c | 330
-b/res/res_pjsip_t38.c | 40
-b/res/res_pjsip_transport_websocket.c | 4
-b/res/res_realtime.c | 56
-b/res/res_resolver_unbound.c | 6
-b/res/res_rtp_asterisk.c | 1100 +
-b/res/res_sorcery_config.c | 14
-b/res/res_sorcery_memory_cache.c | 18
-b/res/res_speech.c | 7
-b/res/res_srtp.c | 8
-b/res/res_stasis.c | 57
-b/res/res_stasis_playback.c | 4
-b/res/res_statsd.c | 35
-b/res/res_stir_shaken.c | 1663 ++
-b/res/res_stir_shaken.exports.in | 6
-b/res/res_stir_shaken/certificate.c | 388
-b/res/res_stir_shaken/certificate.h | 119
-b/res/res_stir_shaken/curl.c | 199
-b/res/res_stir_shaken/curl.h | 73
-b/res/res_stir_shaken/general.c | 286
-b/res/res_stir_shaken/general.h | 111
-b/res/res_stir_shaken/stir_shaken.c | 119
-b/res/res_stir_shaken/stir_shaken.h | 55
-b/res/res_stir_shaken/store.c | 202
-b/res/res_stir_shaken/store.h | 37
-b/res/stasis/app.c | 15
-b/res/stasis/command.c | 2
-b/res/stasis/control.c | 14
-b/res/stasis/messaging.c | 11
-b/res/stasis/stasis_bridge.c | 16
-b/res/stasis/stasis_bridge.h | 3
-b/rest-api-templates/make_ari_stubs.py | 2
-b/rest-api-templates/res_ari_resource.c.mustache | 2
-b/rest-api/api-docs/bridges.json | 9
-b/rest-api/api-docs/channels.json | 143
-b/rest-api/api-docs/endpoints.json | 20
-b/rest-api/resources.json | 2
-b/tests/CI/buildAsterisk.sh | 2
-b/tests/CI/gates.jenkinsfile | 12
-b/tests/CI/periodics-daily.jenkinsfile | 11
-b/tests/CI/publishAsteriskDocs.sh | 4
-b/tests/CI/ref_debug.jenkinsfile | 9
-b/tests/CI/unittests.jenkinsfile | 9
-b/tests/CI/universal-asterisk-nongerrit.jenkinsfile | 452
-b/tests/test_conversions.c | 153
-b/tests/test_data_buffer.c | 2
-b/tests/test_locale.c | 12
-b/tests/test_res_pjsip_session_caps.c | 176
-b/tests/test_res_rtp.c | 516
-b/tests/test_scope_trace.c | 126
-b/tests/test_stasis.c | 2
-b/tests/test_stream.c | 2
-b/tests/test_strings.c | 39
-b/tests/test_taskprocessor.c | 78
-b/tests/test_utils.c | 11
-b/third-party/pjproject/Makefile | 3
-b/third-party/pjproject/configure.m4 | 199
-b/third-party/pjproject/patches/0011-sip_inv_patch.patch | 39
-b/third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch | 39
-b/third-party/pjproject/patches/0050-fix-race-parallel-build.patch | 72
-b/third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch | 28
-b/third-party/pjproject/patches/config_site.h | 4
-b/third-party/pjproject/pjproject-2.10.tar.bz2.md5 | 2
-b/third-party/versions.mak | 2
-b/utils/Makefile | 6
-b/utils/astman.c | 2
-b/utils/db1-ast/hash/ndbm.c | 3
-b/utils/db1-ast/include/db.h | 1
-b/utils/extconf.c | 10
-b/utils/muted.c | 11
-third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
-third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch | 53
-third-party/pjproject/patches/0030-ssl-regression-fix.patch | 105
-third-party/pjproject/patches/0031-transport-regression-fix.patch | 187
-third-party/pjproject/pjproject-2.9.tar.bz2.md5 | 2
-392 files changed, 29541 insertions(+), 9926 deletions(-)</pre><br></html>
\ No newline at end of file
+++ /dev/null
- Release Summary
-
- asterisk-18.0.0
-
- Date: 2020-10-19
-
- <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 is the first release of a major new version of Asterisk. For a list
- of new features that have been included with this release, please see the
- CHANGES file inside the source package. Since this is a new major release,
- users are encouraged to do extended testing before upgrading to this
- version in a production environment.
-
- 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
- 70 Sean Bright 1 tests/test_utils.c. 26 Joshua C. Colp
- 54 George Joseph 23 Alexander Traud
- 47 Joshua C. Colp 13 Kevin Harwell
- 31 Kevin Harwell 9 Ross Beer
- 26 Alexander Traud 8 sungtae kim
- 17 Ben Ford 8 nappsoft
- 16 Joshua Colp 7 Walter Doekes
- 16 Jaco Kroon 7 Torrey Searle
- 11 Walter Doekes 6 Frederic LE FOLL
- 9 sungtae kim 5 Guido Falsi
- 9 Alexander Traud 5 cmaj
- 8 Torrey Searle 5 George Joseph
- 8 Corey Farrell 4 Jaco Kroon
- 7 Richard Mudgett 4 Jean Aunis - Prescom
- 6 Frederic LE FOLL 4 Pascal Cadotte Michaud
- 6 Pirmin Walthert 4 Ross Beer
- 6 Guido Falsi 3 Joshua Elson
- 5 Asterisk Development 3 Sean Bright
- Team 3 Salah Ahmed
- 5 Alexei Gradinari 3 Nickolay V. Shmyrev
- 3 Pascal Cadotte Michaud 3 lvl
- 3 Igor Goncharovsky 3 Alexei Gradinari
- 3 Jean Aunis 2 Stas Kobzar
- 3 Nickolay Shmyrev 2 Michael Neuhauser
- 3 lvl 2 Ruddy G
- 2 Rodrigo RamÃrez 2 Joeran Vinzens
- Norambuena 2 Timothy Vanderaerden
- 2 Andrew Siplas 2 Sébastien Duthil
- 2 Salah Ahmed 2 Peter Sokolov
- 2 Michael Neuhauser 2 Joseph Ades
- 2 cmaj 2 Gregory Massel
- 1 Kevin Reeves 2 Andrew Siplas
- 1 Michael Goryainov 2 Jared Smith
- 1 Università di Bologna - 2 Jonathan Harris
- CESIA VoIP 2 Michael Neuhauser
- 1 Chris Savinovich 1 Ramarajan
- 1 Nathan Bruning 1 Andrey V. T.
- 1 Paulo Vicentini 1 tootai
- 1 Peter Turczak 1 Martin Tomec
- 1 Sungtae Kim 1 AvayaXAsterisk
- 1 Dan Cropp 1 Joshua C. Colp
- 1 Jared Smith 1 Etienne Lessard
- 1 Stas Kobzar 1 Benjamin Keith Ford
- 1 Daniel Heckl 1 Yoooooo Ha
- 1 Dennis Buteyn 1 kevin@phoneburner.com
- 1 Bernard Merindol 1 Juan Martin
- 1 Jonathan Rose 1 Sylvain Afchain
- 1 Kfir Itzhak 1 Speed Dial Dave
- 1 Roger James 1 Andrew Yager
- 1 Sebastian Kemper 1 Jean-Denis Girard
- 1 Christoph Moench-Tegeder 1 Marian Piater
- 1 Boris P. Korzun 1 Bernard Merindol
- 1 Evandro César Arruda 1 Martin Zeh
- 1 Moises Silva 1 Corey Farrell
- 1 Chris-Savinovich 1 Dan Cropp
- 1 Michael Cargile 1 Moises Silva
- 1 Sylvain Afchain 1 Alexey Vasilyev
- 1 Florian Floimair 1 Thomas Johnson
- 1 Nicholas John Koch 1 Seán C. McCord
- 1 Peter Sokolov (License 1 Dirk Wendland
- #7070) 1 Bryan Nelson
- 1 Martin Tomec 1 Sam Banks
- 1 Thomas Arimont (license 1 Misha Vodsedalek
- 5525) 1 Nicholas John Koch
- 1 Seán C McCord 1 Richard Kenner
- 1 Patrick Verzele 1 EDV O-TON
- 1 snuffy 1 Byron Clark
- 1 Sebastien Duthil 1 Christoph
- 1 Jason Hord (license Moench-Tegeder
- 6978) 1 sstream
- 1 Dmitriy Serov
- 1 Alex
- 1 candrews
- 1 Sébastien Duthil
- 1 Robert Sutton
- 1 Evandro César Arruda
- 1 Paul Brooks
- 1 Yury Kirsanov
- 1 Jason Hord
- 1 Michael Cargile
- 1 Kevin Flyn
- 1 Shlomi Gutman
- 1 George Joseph
- 1 Frank Matano
- 1 Cédric Bassaget
- 1 Dan Jenkins
- 1 Jim Van Meggelen
- 1 Patrick Wakano
- 1 Jeremiah Gadd
- 1 Michael
- 1 Daniel Heckl
- 1 Boris P. Korzun
- 1 Kilburn
- 1 Bernhard Schmidt
- 1 Alexander Traud
- 1 Joeran Vinzens
- 1 Dennis
- 1 Vitold
- 1 Anton Satskiy
- 1 Kevin Flyn
- 1 David Cunningham
- 1 Jim Van Meggelen
- 1 Vitold
- 1 Florian Floimair
- 1 Robert Sutton
- 1 Daniel
- 1 Dan Jenkins
- 1 Ove Aursand
- 1 Dmitry Wagin
- 1 Robin Leffmann
- 1 Mitch Claborn
- 1 Jonathan Hunter
- 1 Joshua Roys
- 1 Olivier Krief
- 1 Paul Brooks
- 1 Maciej Michno
- 1 Kevin Reeves
- 1 Niklas Larsson
- 1 Bernhard Schmidt
- 1 Christoph
- Moench-Tegeder
- 1 Maciej Michno
- 1 Stas Kobzar
- 1 Cedric BASSAGET
- 1 EDV O-TON
- 1 Ted G
- 1 Frank Matano
- 1 Yury Kirsanov
- 1 Anton Satskiy
- 1 David M. Lee
- 1 Patrick Wakano
- 1 Michael Goryainov
- 1 Niklas Larsson
- 1 Sebastian Kemper
- 1 Francois Blackburn
- 1 Università di Bologna
- - CESIA VoIP
- 1 Richard Kenner
- 1 Niksa Baldun
- 1 Ian Jones
- 1 Jean-Denis Girard
- 1 Dmitriy Serov
- 1 Peter Turczak
- 1 Roger James
- 1 Paulo Vicentini
- 1 Ted G
- 1 Martin Zeh
- 1 Università di Bologna
- - CESIA VoIP
- 1 Marin Odrljin
- 1 Jonas Swiatek
- 1 Eliel Sardañons
- 1 AvayaXAsterisk
- 1 Dirk Wendland
- 1 Joshua Roys
- 1 Mark
- 1 Dan Cropp
- 1 Jonathan Harris
- 1 Matt Addison
- 1 Leandro Dardini
- 1 alex
- 1 Chris Savinovich
- 1 xrobau
- 1 David Lee
- 1 Nicholas John Koch
- 1 Peter Sokolov
- 1 Eliel Sardañons
- 1 Sean Bright
- 1 Aheliotech
- 1 Bill Kervaski
- 1 Cyril Ramière
- 1 Jørgen H
- 1 Niksa Baldun
- 1 Kfir Itzhak
-
- ----------------------------------------------------------------------
-
- 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.
- * [4a1cadeadb] 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
- * [7e3a6e158f] 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
- * [18f5f5fc99] Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add
- NULL checks before using session media
-
- New Feature
-
- Category: Applications/app_senddtmf
-
- ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
- of only "sending"
- Reported by: lvl
- * [772b59034f] lvl -- app_senddtmf: Add receive mode to AMI Action
- PlayDTMF
-
- Category: Core/General
-
- ASTERISK-6863: [patch] allow Asterisk to set high ToS bits as non-root on
- Linux
- Reported by: Matt Addison
- * [a107e85b2e] Alexander Traud -- install_prereq: Add libcap for high
- bits in DiffServ/ToS.
-
- Category: Core/Jitterbuffer
-
- ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
- Reported by: Joshua C. Colp
- * [7298a785ad] Joshua Colp -- func_jitterbuffer: Add audio/video sync
- support.
-
- Category: Functions/func_curl
-
- ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs"
- doesn't do anything
- Reported by: candrews
- * [0c2bf1664c] Sean Bright -- func_curl: Add 'followlocation' option to
- CURLOPT()
- ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
- Reported by: Martin Tomec
- * [d257a0898e] 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
- * [cf364cd007] sungtae kim -- res_musiconhold: Added unregister realtime
- moh class
-
- Category: Resources/res_pjsip_endpoint_identifier_ip
-
- ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on
- source port
- Reported by: Sean Bright
- * [312abaa1fe] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add
- port matching support
-
- Category: pjproject/pjsip
-
- ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup
- From header URI domain
- Reported by: Stas Kobzar
- * [c7270dca81] Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN
-
- Bug
-
- Category: .Release/Targets
-
- ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register
- Reported by: Chris Savinovich
- * [172e183b9d] Kevin Harwell -- res_pjsip_mwi: add better handling of
- solicited vs unsolicited subscriptions
-
- Category: Applications/General
-
- ASTERISK-28954: StreamEcho() only returns 1 active stream
- Reported by: Bill Kervaski
- * [00a52b4752] Joshua C. Colp -- app_stream_echo: Fix state of added
- streams.
- ASTERISK-16676: DAHDIRAS fails to properly initiate pppd unless asterisk
- is running as root
- Reported by: Jaco Kroon
- * [4f92dcd66b] Jaco Kroon -- dahdiras: Only set plugin dahdi.so to pppd
- if we're running as root.
-
- Category: Applications/app_amd
-
- ASTERISK-28608: app_amd: Use time calculation to calculate timeout
- Reported by: Michael Cargile
- * [5bda460300] Michael Cargile -- app_amd: Fixed timeout issue
-
- Category: Applications/app_chanisavail
-
- ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to
- deactivate CDR.
- Reported by: Frederic LE FOLL
- * [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail
- sometimes fails to deactivate CDR.
- ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in
- cdr.conf
- Reported by: Frederic LE FOLL
- * [2d0eee5418] Frederic LE FOLL -- ChanIsAvail() generates a CDR when
- unanswered=yes in cdr.conf.
-
- Category: Applications/app_confbridge
-
- ASTERISK-28841: app_confbridge: Add support for disabling text messaging
- for a user
- Reported by: Joshua C. Colp
- * [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling
- text messaging.
- ASTERISK-28790: Crash during conference call using confbridge and video
- Reported by: Pascal Cadotte Michaud
- * [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient
- space for worst case NACK.
-
- Category: Applications/app_fax
-
- ASTERISK-28848: app_fax: Compile.
- Reported by: Alexander Traud
- * [26b8c99963] Alexander Traud -- app_fax: SpanDSP headers do not use
- ast_malloc; ignore that.
-
- Category: Applications/app_meetme
-
- ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
- 17.0.0
- Reported by: George Joseph
- * [ed394ce5b1] Joshua C. Colp -- configure: Add check for MySQL client
- bool and my_bool type usage.
- * [a47cb71bb1] George Joseph -- Build: Fix compile issues with seldom
- used modules
-
- Category: Applications/app_mixmonitor
-
- ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between
- AMI MixMonitor and hangup
- Reported by: Joshua C. Colp
- * [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to
- attach to hung up channels.
-
- Category: Applications/app_osplookup
-
- ASTERISK-28804: [patch] app_osplookup.c: Avoid a format truncation.
- Reported by: Alexander Traud
- * [527e4f6542] Alexander Traud -- app_osplookup: Avoid a format
- truncation.
-
- Category: Applications/app_queue
-
- ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
- Reported by: Ove Aursand
- * [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a
- call as abandoned
- ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned
- Reported by: Kfir Itzhak
- * [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a
- call as abandoned
- ASTERISK-29034: Lastpause of realtime members is reseting
- Reported by: Evandro César Arruda
- * [36dd15c659] Evandro César Arruda -- app_queue: Member lastpause time
- reseting
- ASTERISK-28951: Inconsistent behaviour queues.conf when there is (not) a
- [general] section
- Reported by: Walter Doekes
- * [312c23b0e1] Walter Doekes -- app_queue: (Breaking change)
- shared_lastcall and autofill default to no
- ASTERISK-28950: Stale code in app_queue to check untouched channel
- Reported by: Walter Doekes
- * [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in
- try_calling
- ASTERISK-28644: Stale comment in app_queue about ring_entry exception
- Reported by: Walter Doekes
- * [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in
- try_calling
- * [0e750cdd10] Walter Doekes -- app_queue: Fix old confusing comment
- about when the members are called
- ASTERISK-28952: Queue wrapuptime sometimes not respected (based on stale
- lastcall time)
- Reported by: Walter Doekes
- * [0fb6738314] Walter Doekes -- app_queue: Read latest wrapuptime
- instead of (possibly stale) copy
- ASTERISK-28829: app_queue: leaking stasis subscription when Redirecting
- call
- Reported by: lvl
- * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in
- app_queue to avoid leaked stasis subscriptions
- ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
- Reported by: Etienne Lessard
- * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in
- app_queue to avoid leaked stasis subscriptions
- ASTERISK-28349: Pause reason not reported in QueueMember AMI event
- Reported by: Niksa Baldun
- * [9522390a69] Sean Bright -- app_queue: Deprecate the
- QueueMemberPause.Reason field
-
- Category: Applications/app_record
-
- ASTERISK-28682: app_record: Lack of `beep` audio file causes application
- to return error and hangup
- Reported by: Corey Farrell
- * [2f8b20b949] Corey Farrell -- app_record: Do not hang up if beep audio
- is missing
-
- Category: Applications/app_system
-
- ASTERISK-28776: Non async-signal-safe syscalls used after fork before exec
- Reported by: nappsoft
- * [6b2d945174] Pirmin Walthert -- app.c: make sure that no
- non-async-signal-safe syscalls are used after
-
- Category: Applications/app_voicemail
-
- ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it
- is not sent by email/processed by the mailcmd command
- Reported by: Leandro Dardini
- * [b575868000] Sean Bright -- app_voicemail: Process urgent messages
- with mailcmd
- ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage
- enabled and realtime voicemail_data is used
- Reported by: Stas Kobzar
- * [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving
- message with realtime voicemail
- ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail
- to leave message
- Reported by: Jim Van Meggelen
- * [9be89d9913] Sean Bright -- app_voicemail: Set globals to default
- values when voicemail.conf missing
-
- Category: Applications/app_voicemail/IMAP
-
- ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because
- not checking mailstream
- Reported by: Alexei Gradinari
- * [15624d9a7a] Alexei Gradinari -- app_voicemail/IMAP: check mailstream
- not NULL in leave_voicemail
-
- Category: Applications/app_voicemail/ODBC
-
- ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage
- enabled and realtime voicemail_data is used
- Reported by: Stas Kobzar
- * [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving
- message with realtime voicemail
-
- Category: Bridges/bridge_builtin_features
-
- ASTERISK-28920: bridge show all causes crash
- Reported by: sungtae kim
- * [25ae412f75] sungtae kim -- bridge.c: Fixed null pointer exception
-
- 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
- * [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
- directmedia for ptime other than default ptime.
-
- Category: Bridges/bridge_softmix
-
- ASTERISK-28944: bridge_softmix: Transitioning a stream from inactive ->
- sendrecv/sendonly doesn't re-negotiation
- Reported by: Joshua C. Colp
- * [8ad06394c4] Joshua C. Colp -- bridge_softmix: Add additional old
- states for adding new source.
- ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp
- packets
- Reported by: Jonathan Hunter
- * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio
- from mixed frame.
- ASTERISK-28819: [patch] bridge_softmix_binaural: Show state in menuselect.
- Reported by: Alexander Traud
- * [7febd22304] Alexander Traud -- bridge_softmix_binaural: Show state in
- menuselect.
- ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix
- bridge
- Reported by: Kevin Harwell
- * [e77cb32583] Kevin Harwell -- bridge_softmix: clear hold when joining
- a softmix bridge
-
- Category: CDR/General
-
- ASTERISK-28677: CDR billsec is always 0 for transferred calls
- Reported by: Maciej Michno
- * [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when
- leaving a parking bridge
- ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to
- deactivate CDR.
- Reported by: Frederic LE FOLL
- * [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail
- sometimes fails to deactivate CDR.
- ASTERISK-28566: CDR backend unload problem during active call(s)
- Reported by: Marian Piater
- * [51850a79ef] 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
- * [52ade18420] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql
- res_config_pgsql: compatibility with PostgreSQL 12
-
- Category: Channels/chan_dahdi
-
- ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times
- out after 0:16:40
- Reported by: Andrew Siplas
- * [5bd7281442] Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to
- better reflect "no timeout"
- ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after
- a short alarm
- Reported by: Frederic LE FOLL
- * [a68299f508] 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
- * [4072e219f7] 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
- * [41b67f150e] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource)
- when a PRI channel hangs up
-
- Category: Channels/chan_local
-
- ASTERISK-28938: core_unreal / core_local: Add support for multistream and
- re-negotiation
- Reported by: Joshua C. Colp
- * [de2813cf23] Joshua C. Colp -- core_unreal / core_local: Add
- multistream and re-negotiation.
- ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
- Reported by: Etienne Lessard
- * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in
- app_queue to avoid leaked stasis subscriptions
-
- Category: Channels/chan_pjsip
-
- ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
- Reported by: Joseph Ades
- * [31fbfc5e95] Kevin Harwell -- chan_pjsip: disallow
- PJSIP_SEND_SESSION_REFRESH pre-answer execution
- * [4eba6b9eb2] Kevin Harwell -- PJSIP_MEDIA_OFFER: override
- configuration on refresh
- ASTERISK-28886: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject
- 2.7.2
- Reported by: Jared Smith
- * [8b925fbda3] Kevin Harwell -- chan_pjsip: don't use PJSIP_SC_NULL as
- it only exists pjproject 2.8+
- ASTERISK-28923: T.38 Segfaults in chan_pjsip_queryoption
- Reported by: Yury Kirsanov
- * [41f3a7da4d] George Joseph -- res_fax: Don't start a gateway if either
- channel is hung up
- ASTERISK-28835: IPv6 addresses in SDP incorrectly formatted
- Reported by: Daniel Heckl
- * [9f117ac9ef] Daniel Heckl -- res_pjsip: Fixed format of IPv6 addresses
- for external media addresses
- ASTERISK-28817: chan_pjsip: constant DTMF tone if RTP is not setup yet
- Reported by: Kevin Harwell
- * [fa3c8f94e0] Kevin Harwell -- chan_pjsip: digit_begin - constant DTMF
- tone if RTP is not setup yet
- ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during
- direct-media (native_rtp) bridge
- Reported by: Michael Neuhauser
- * [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore
- rtptimeout if direct-media is active
- ASTERISK-28759: A non negotiated rtp frame causes call disconnection when
- there is a SSRC change
- Reported by: Paulo Vicentini
- * [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when
- remote SSRC changes.
- ASTERISK-28766: PJSIP blind transfer not completed after using
- Proceeding()
- Reported by: lvl
- * [d1a2ff0aaf] lvl -- res_pjsip_refer: ensure refer progress is still
- sent after Proceeding()
- ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables"
- field
- Reported by: Jean Aunis - Prescom
- * [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text
- message variables work in ARI
- ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group
- endpoint/call_group
- Reported by: Jean-Denis Girard
- * [b40dd11afe] Sean Bright -- res_pjsip_config_wizard: Fix change
- detection for wizard settings
- ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response
- Contact
- Reported by: Ross Beer
- * [cbc1136704] George Joseph -- res_pjsip_nat: Restore original contact
- for REGISTER responses
- ASTERISK-28578: race condition on pjsip channelstats command
- Reported by: Salah Ahmed
- * [ddb0091da5] Salah Ahmed -- Crash during "pjsip show channelstats"
- execution
- ASTERISK-28561: Asterisk Deadlocks
- Reported by: Aheliotech
- * [bf6f27388d] 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
- * [c03f50c1c8] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
- on hungup channel
- ASTERISK-28538: chan_pjsip: Deadlock on fax detection
- Reported by: Joshua C. Colp
- * [c358da472e] Joshua Colp -- chan_pjsip: Relock correct channel during
- "fax" redirect.
-
- Category: Channels/chan_sip/General
-
- ASTERISK-29011: chan_sip: ToHost property not cleared on reload
- Reported by: Dennis
- * [9058d9e591] Dennis Buteyn -- chan_sip: Clear ToHost property on peer
- when changing to dynamic host
- ASTERISK-28957: chan_sip: chan_sip does not process 400 response to an
- INVITE.
- Reported by: Frederic LE FOLL
- * [a423f935c9] Frederic LE FOLL -- chan_sip: chan_sip does not process
- 400 response to an INVITE.
- ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp
- packets
- Reported by: Jonathan Hunter
- * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio
- from mixed frame.
- ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
- Reported by: Jaco Kroon
- * [365d007eb6] Jaco Kroon -- chan_sip: in case of tcp/tls, be less
- annoying about tx errors.
- ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
- Reported by: Jean Aunis - Prescom
- * [9c9296c635] Jean Aunis -- chan_sip: voice frames are no longer
- transmitted after emitting a COLP
- ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
- check failure when negociated ptime is not default ptime.
- Reported by: Frederic LE FOLL
- * [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
- directmedia for ptime other than default ptime.
-
- Category: Channels/chan_sip/Interoperability
-
- ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should
- return 503
- Reported by: Walter Doekes
- * [43620cbf6c] Walter Doekes -- chan_sip: Return 503 if we're out of RTP
- ports
- ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed:
- no audio
- Reported by: Walter Doekes
- * [711a3fed56] Walter Doekes -- chan_sip: Always process updated SDP on
- media source change
-
- Category: Channels/chan_sip/Messaging
-
- ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears
- empty in the dialplan
- Reported by: Frank Matano
- * [f309b86e36] Sean Bright -- chan_sip.c: Stop handling continuation
- lines after reading headers
-
- Category: Channels/chan_sip/TCP-TLS
-
- ASTERISK-28372: Asterisk REPLY Wrong Contact header port (TCP)
- Reported by: Anton Satskiy
- * [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with
- non-default TCP/TLS ports.
- ASTERISK-24428: Document that Asterisk will use the default SIP ports
- (5060 for TCP, 5061 for TLS) if the extern option variants aren't used
- Reported by: sstream
- * [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with
- non-default TCP/TLS ports.
- ASTERISK-27195: chan_sip: only sets ToS bits on UDP socket, ignoring TCP
- and TLS sockets
- Reported by: Joshua Roys
- * [4d0ab620be] Alexander Traud -- chan_sip: DiffServ/ToS not only on UDP
- but also on TCP and TLS sockets.
-
- Category: Channels/chan_sip/Transfers
-
- ASTERISK-28677: CDR billsec is always 0 for transferred calls
- Reported by: Maciej Michno
- * [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when
- leaving a parking bridge
-
- Category: Channels/chan_unistim
-
- ASTERISK-28803: [patch] chan_unistim: Avoid tautological warnings with
- clang.
- Reported by: Alexander Traud
- * [b38f664250] Alexander Traud -- chan_unistim: Avoid tautological
- warnings with clang.
- ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at
- end of a struct
- Reported by: Alexander Traud
- * [3863ab9af9] 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
- * [e4289b9e56] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is
- defined when necessary
- * [b096389660] Sean Bright -- codec_resample: Upgrade speex_resample to
- fix up-sampling bug
-
- Category: Codecs/codec_silk
-
- ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation'
- output
- Reported by: Sean Bright
- * [dfad69ce7c] Sean Bright -- translate.c: Fix silk 24kHz truncation in
- 'core show translation'
-
- Category: Configs/Basic-PBX
-
- ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order
- mark is present
- Reported by: Robin Leffmann
- * [40b5cf8f52] Sean Bright -- config.c: Skip UTF-8 BOMs if present when
- reading config files
-
- Category: Configs/Samples
-
- ASTERISK-29123: logger.conf.sample missing comment mark on line 115
- Reported by: Andrew Siplas
- * [79d749d2b5] Andrew Siplas -- logger.conf.sample: add missing comment
- mark
-
- Category: Contrib/General
-
- ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed
- to due to invalid syntax
- Reported by: Richard Kenner
- * [095c204fe0] snuffy -- contrib/valgrind: Fix use of frame-level
- suppression
- ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
- Reported by: Pascal Cadotte Michaud
- * [e494d5fd76] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid
- typo
-
- Category: Core/ACL
-
- ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault
- on reading rule from realtime
- Reported by: Andrew Yager
- * [7a43bedd72] Sean Bright -- acl.c: Coerce a NULL pointer into the
- empty string
-
- Category: Core/Bridging
-
- ASTERISK-28841: app_confbridge: Add support for disabling text messaging
- for a user
- Reported by: Joshua C. Colp
- * [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling
- text messaging.
-
- Category: Core/BuildSystem
-
- ASTERISK-28929: pjproject_bundled: Honor --without-pjproject.
- Reported by: Alexander Traud
- * [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor
- --without-pjproject.
- ASTERISK-28837: pjproject_bundled: Honor --without-pjproject.
- Reported by: Alexander Traud
- * [966acc6251] Alexander Traud -- pjproject_bundled: Honor
- --without-pjproject.
- ASTERISK-28824: BuildSystem: Search for Python/C API when possibly needed
- only.
- Reported by: Alexander Traud
- * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API
- when possibly needed only.
- ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming
- Language is python-2.7.
- Reported by: Alexander Traud
- * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API
- when possibly needed only.
- ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
- Reported by: Alexander Traud
- * [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and
- doc/pdf leftovers.
- ASTERISK-28818: [patch] BuildSystem: Allow space in path.
- Reported by: Alexander Traud
- * [7a04947abd] Alexander Traud -- BuildSystem: Allow space in path.
- ASTERISK-28487: compile menuselect on gentoo
- Reported by: Kilburn
- * [e40f248fac] Sean Bright -- menuselect: Fix curses build on Gentoo
- Linux
-
- Category: Core/Channels
-
- ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
- Reported by: Etienne Lessard
- * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in
- app_queue to avoid leaked stasis subscriptions
- ASTERISK-28795: channel: write to a stream on multi-frame writes
- Reported by: Kevin Harwell
- * [3c345ec56d] Kevin Harwell -- channel: write to a stream on
- multi-frame writes
- ASTERISK-28499: translate: Crash when frame does not have a "src" field
- set
- Reported by: Gregory Massel
- * [1e9714a050] Joshua Colp -- AST-2019-005 - translate: Don't assume all
- frames will have a src.
-
- Category: Core/Configuration
-
- ASTERISK-28955: "setvar" doesn't work properly in dahdi-channels.conf
- Reported by: Marin Odrljin
- * [d88e230037] Guido Falsi -- chan_dadhi: Fix setvar in dahdi channels
- ASTERISK-23756: setvar directive when used in template and a child of said
- template, results in duplicate variable names
- Reported by: Michael Goryainov
- * [32ce6e9a06] Michael Goryainov -- channels: Allow updating variable
- value
-
- Category: Core/General
-
- ASTERISK-28797: [patch] tcptls: Fix notice when TLS is enabled but not
- configured.
- Reported by: Alexander Traud
- * [f9ea75d117] Alexander Traud -- tcptls: Fix notice when TLS is enabled
- but not supported.
- ASTERISK-28839: Sporadic crashes with Segmentation fault
- Reported by: Joeran Vinzens
- * [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP
- re-negotiation scenarios.
- ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between
- AMI MixMonitor and hangup
- Reported by: Joshua C. Colp
- * [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to
- attach to hung up channels.
- ASTERISK-28498: cel / cdr: Event times may be incorrect
- Reported by: Joshua C. Colp
- * [261646c1c4] 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
- * [3656c42cb0] Kevin Harwell -- various modules: json integer overflow
-
- Category: Core/Stasis
-
- ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables"
- field
- Reported by: Jean Aunis - Prescom
- * [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text
- message variables work in ARI
- ASTERISK-28553: stasis.c: Crash during unload
- Reported by: Kevin Harwell
- * [729b286d59] Joshua Colp -- stasis: Pass bumped topic_all reference to
- proxy_dtor.
-
- Category: Core/Streams
-
- ASTERISK-28870: streams: One memory leak and one issue cloning streams
- Reported by: George Joseph
- * [7fbfbe7da0] George Joseph -- streams: Fix one memory leak and one
- formats ref issue
- ASTERISK-28846: stream: Enforce formats immutability
- Reported by: Joshua C. Colp
- * [1c5e68580a] Joshua C. Colp -- stream: Enforce formats immutability
- and ensure formats exist.
- ASTERISK-28625: Playback of local files impacted by large media cache
- Reported by: Kevin Reeves
- * [c626ccec12] Kevin Reeves -- main/file.c: Limit media cache usage to
- remote files.
-
- Category: Core/UDPTL
-
- ASTERISK-28483: packet lost on UDPTL wrap around
- Reported by: Torrey Searle
- * [084901d548] Torrey Searle -- main/udptl.c: correctly handle udptl
- sequence wrap around
-
- Category: Documentation
-
- ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
- Reported by: Alexander Traud
- * [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and
- doc/pdf leftovers.
- ASTERISK-24484: Update documentation for statsd module - usage
- requirements unclear
- Reported by: Dan Jenkins
- * [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does
- nothing on its own
- ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for
- hostnames
- Reported by: Joshua C. Colp
- * [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
- support for hostnames
- ASTERISK-28507: Wiki docs missing for MessageWaiting
- Reported by: David M. Lee
- * [d5f3ec92d0] George Joseph -- CI: Update buildAsterisk.sh to do a
- "make full"
-
- Category: Functions/General
-
- ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
- Reported by: Pascal Cadotte Michaud
- * [bf4dd3d837] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
- argument documentation
- * [7e3015d779] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
- argument documentation
- ASTERISK-26481: FILE function grabs garbage along with read data when
- target line has no newline
- Reported by: Jonathan Harris
- * [bf7c808604] Sean Bright -- func_env: Prevent FILE() from reading
- garbage at end-of-file
-
- Category: Functions/func_aes
-
- ASTERISK-28788: func_aes: incorrectly printing error 'declined to load'
- Reported by: Alexander Traud
- * [cd8cbf7384] Alexander Traud -- func_aes: Avoid incorrect error
- message on load.
-
- Category: Functions/func_channel
-
- ASTERISK-28796: func_channel: cannot read fields exten, context,
- userfield, channame from dialplan
- Reported by: Sébastien Duthil
- * [d40e343710] Sebastien Duthil -- func_channel: allow reading 4 fields
- from dialplan
-
- Category: Functions/func_enum
-
- ASTERISK-26711: func_enum: ENUM code wrong case
- Reported by: Vitold
- * [517224ce85] Sean Bright -- enum.c: Add support for regular expression
- flag in NAPTR record
- ASTERISK-19460: [patch] Function TXTCIDNAME never actually makes DNS calls
- and always returns an empty string
- Reported by: George Joseph
- * [ab63f0cd0f] Sean Bright -- enum.c: Make ast_get_txt() actually do
- something.
-
- Category: Functions/func_odbc
-
- ASTERISK-20325: Comments in configs/func_odbc.conf.sample are not
- consistent with examples. Missing examples.
- Reported by: Olivier Krief
- * [c4e0983742] Sean Bright -- func_odbc.conf.sample: Clarify sample
- documentation
- ASTERISK-28497: func_odbc: truncating Unicode string on readsql
- Reported by: Boris P. Korzun
- * [8979921da9] Boris P. Korzun -- func_odbc: acf_odbc_read() and
- cli_odbc_read() unicode support
-
- Category: Functions/func_version
-
- ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
- versions
- Reported by: cmaj
- * [543f936147] cmaj -- Makefile: Fix certified version numbers
-
- Category: General
-
- ASTERISK-28930: ./configure --without-ssl build failure
- Reported by: Jaco Kroon
- * [9b5042433b] Joshua C. Colp -- menuselect: Resolve infinite loop in
- dependency scenario.
- ASTERISK-28838: AST_MODULE_INFO requires, MODULEINFO does not mention
- Reported by: Alexander Traud
- * [abf4d74384] Alexander Traud -- cdr_odbc: Sync load- and build-time
- deps.
- * [191f136260] Alexander Traud -- res_pjsip_refer: Add build-time
- dependency.
- * [5c2b8fdeca] Alexander Traud -- app_getcpeid: Add build-time
- dependency.
- * [008f46bf1e] Alexander Traud -- res_pjsip: Sync load- and build-time
- deps.
- * [e2affa3b0a] Alexander Traud -- curl: Add build-time dependency.
- * [f1135b453b] Alexander Traud -- res_pjsip: Add build-time dependency.
- ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
- Reported by: Ted G
- * [39c920ac78] George Joseph -- res_rtp_asterisk: Add frame list
- cleanups to ast_rtp_read
- ASTERISK-28590: utils.c throws repeated warnings;
- "pthread_attr_setstacksize: Invalid argument"
- Reported by: Speed Dial Dave
- * [a4222614c4] 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
- * [a4caaef64c] 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
- * [b805e1237d] Kevin Harwell -- srtp: Fix possible race condition, and
- add NULL checks
-
- Category: PBX/General
-
- ASTERISK-29046: pbx: Deadlock when doing a reload, while simultaneously
- doing an ExtensionState on a pattern match hint that ends up adding an
- extension
- Reported by: Ramarajan
- * [6d50d152d8] Joshua C. Colp -- pbx: Fix hints deadlock between reload
- and ExtensionState.
- ASTERISK-28695: core: minmemfree watermark uses free RAM, not available
- RAM
- Reported by: Kevin Flyn
- * [50d02d6194] Sean Bright -- pbx.c: Include filesystem cache in free
- memory calculation
- ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent
- command pri show span X
- Reported by: Dirk Wendland
- * [ee7d72eb72] George Joseph -- sig_pri: Fix deadlock caused by
- sig_pri_queue_hangup
-
- Category: PBX/pbx_config
-
- ASTERISK-28534: Segmentation fault when there is no priority for an
- extension
- Reported by: Timothy Vanderaerden
- * [702019fc80] Sean Bright -- pbx: Prevent Realtime switch crash on
- invalid priority
-
- Category: PBX/pbx_dundi
-
- ASTERISK-21205: [patch] dundi_read_result crash due to negative number
- Reported by: Jaco Kroon
- * [40e93b0240] Jaco Kroon -- dundi: fix NULL dereference.
-
- Category: Resources/res_ari
-
- ASTERISK-28948: ARI channel create doesn't referencing the channel_id
- parameter
- Reported by: sungtae kim
- * [bbe0f2230d] sungtae kim -- res_ari: Fix create channel request
- channelId parameter parsing
- ASTERISK-28679: stasis application is destroyed after its creation
- Reported by: Francois Blackburn
- * [4206830a52] Kevin Harwell -- res_stasis: trigger cleanup after update
- ASTERISK-28585: ari/resource_events: Crash in event session cleanup
- Reported by: Kevin Harwell
- * [360936ead5] Joshua Colp -- res_ari_events: Add module reference when
- a WebSocket is open.
-
- Category: Resources/res_ari_bridges
-
- ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp
- packets
- Reported by: Jonathan Hunter
- * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio
- from mixed frame.
-
- Category: Resources/res_ari_channels
-
- ASTERISK-28940: /channels/create doesn't get any parameters from the body
- Reported by: sungtae kim
- * [fa7c69f40f] sungtae kim -- res_ari: Fix create request body parameter
- parsing.
- ASTERISK-28847: ARI channels cuts the endpoint string over 80 characters
- Reported by: sungtae kim
- * [9ad3d2829c] sungtae kim -- res_ari_channels: Fixed endpoint 80
- characters limit
-
- Category: Resources/res_calendar_exchange
-
- ASTERISK-28572: Memory leaks in res_calendar_exchange and
- res_calendar_icalendar
- Reported by: Yoooooo Ha
- * [16e668c7dd] 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
- * [16e668c7dd] Sean Bright -- res_calendar: Resolve memory leak on
- calendar destruction
-
- Category: Resources/res_corosync
-
- ASTERISK-28888: res_corosync: causes asterisk crash in huge distributed
- environment.
- Reported by: Università di Bologna - CESIA VoIP
- * [0c1c386634] Università di Bologna - CESIA VoIP -- res_corosync: Fix
- crash in huge distributed environment.
-
- Category: Resources/res_fax
-
- ASTERISK-28900: res_fax: Double frame free when gateway in use with
- off-nominal format usage
- Reported by: Gregory Massel
- * [d2500c6273] Joshua C. Colp -- res_fax: Don't consume frames given to
- fax gateway on write.
- ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config
- option
- Reported by: Kevin Harwell
- * [b6f5607359] Kevin Harwell -- res_fax: wrap v21 detected Asterisk
- initiated negotiation with config option
-
- Category: Resources/res_http_websocket
-
- ASTERISK-28975: res_http_websocket: Text payload data doesn't necessary
- include trailing zero
- Reported by: Nickolay V. Shmyrev
- * [e4d24f5137] Nickolay Shmyrev -- res_http_websocket: Avoid reading
- past end of string
- ASTERISK-28562: SIP WSS message not processed until next frame arrives
- Reported by: Robert Sutton
- * [87110c1bdf] Sean Bright -- websocket: Consider pending SSL data when
- waiting for socket input
-
- Category: Resources/res_musiconhold
-
- ASTERISK-28927: Asterisk crash in music on hold
- Reported by: David Cunningham
- * [57554c2834] Sean Bright -- res_musiconhold.c: Prevent crash with
- realtime MoH
- ASTERISK-28892: res_musiconhold: Module res_musiconhold throws false
- warning
- Reported by: Nicholas John Koch
- * [fef97a9a72] Nicholas John Koch -- res_musiconhold: Added check for
- dot character in path of playlist entries to avoid warnings
- ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN
- Reported by: Ross Beer
- * [aeff1f2c53] Sean Bright -- res_musiconhold: Avoid spurious warning
- when 'format' is the empty string
-
- Category: Resources/res_parking
-
- ASTERISK-29042: res_parking: Parker UUID is no longer copied
- Reported by: Misha Vodsedalek
- * [4f0766dcda] Joshua C. Colp -- parking: Copy parker UUID as well.
- ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the
- same
- Reported by: Ross Beer
- * [811ae88da4] 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
- * [807a70b7ae] Joshua Colp -- parking: Fix case where we can't get the
- parker.
- * [e924c5107c] Joshua Colp -- parking: Use channel snapshot instead of
- channel.
-
- Category: Resources/res_pjsip
-
- ASTERISK-28995: res_pjsip_registrar: Expires on statically configured
- contacts is not correct
- Reported by: tootai
- * [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an
- expiration for static contacts.
- ASTERISK-28965: res_pjsip: Apply outbound proxy to static contacts on AOR
- Reported by: Joshua C. Colp
- * [4f86118bd8] Joshua C. Colp -- res_pjsip: Apply AOR outbound proxy to
- static contacts.
- ASTERISK-28936: res_pjsip: crash when dialing non-sip uri
- Reported by: Walter Doekes
- * [e74dde5100] Walter Doekes -- pjsip: Prevent invalid memory access
- when attempting to contact a non-sip URI
- ASTERISK-28794: res_pjsip: Crash when escaping during URI printing
- Reported by: nappsoft
- * [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing
- the contact_user value.
- ASTERISK-26780: res_pjsip: PJSIP Registration Fails when
- transport=transport-udp6
- Reported by: Peter Sokolov
- * [c8dec423d2] Peter Sokolov -- pjsip_resolver.c: Ensure AAAA dns
- requests are made.
- ASTERISK-28854: SIGSEGV when pjsip show history encounters IPV6 address
- Reported by: Roger James
- * [4a072c4890] Roger James -- res_pjsip_history.c: Fix to stop SIGSEGV
- when IPv6 addresses are encountered.
- ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint
- synchronization for a specific AOR
- Reported by: Jason Hord
- * [d845464c76] Jason Hord -- res_pjsip: Don't set endpoint to
- unavailable in all cases.
- ASTERISK-28790: Crash during conference call using confbridge and video
- Reported by: Pascal Cadotte Michaud
- * [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient
- space for worst case NACK.
- ASTERISK-28743: Asterisk is crashing if the 200 OK with SDP
- Reported by: sungtae kim
- * [8147f43756] Sungtae Kim -- res_pjsip_session: Fixed wrong session
- termination
- ASTERISK-23407: Fix the FSF address in the headers of lots of pjproject
- files
- Reported by: Jared Smith
- * [0a7fe3097f] Jared Smith -- indications.conf.sample: Add indication
- tones for Indonesia
- ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop
- Calls
- Reported by: Paul Brooks
- * [313189aae2] Sean Bright -- chan_pjsip: Ignore RTP that we haven't
- negotiated
- ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR
- points to a not existent AOR
- Reported by: Ross Beer
- * [b1be06df8d] 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
- * [377d7bdab6] Sean Bright -- res_pjsip_transport_websocket: Don't put
- brackets around local_name if IPv6
- ASTERISK-28521: pjsip: Memory Leak
- Reported by: Mark
- * [cc83e76aa5] 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
- * [86452c9fa4] Joshua Colp -- res_pjsip: Fix multiple of the same
- contact in "pjsip show contacts".
-
- Category: Resources/res_pjsip_acl
-
- ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed
- Reported by: Timothy Vanderaerden
- * [d6712790cd] Joshua C. Colp -- pjsip: Update ACLs on named ACL
- changes.
-
- Category: Resources/res_pjsip_diversion
-
- ASTERISK-29001: chan_pjsip does not process or forward 181 responses
- Reported by: Torrey Searle
- * [addd295cda] Torrey Searle -- res_pjsip_diversion: handle 181
-
- Category: Resources/res_pjsip_endpoint_identifier_ip
-
- ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for
- hostnames
- Reported by: Joshua C. Colp
- * [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
- support for hostnames
-
- Category: Resources/res_pjsip_logger
-
- ASTERISK-28932: res_pjsip_logger writing too big packets
- Reported by: nappsoft
- * [e8c6e9ae5d] Pirmin Walthert -- res_pjsip_logger: use the correct
- pointer when logging tx_messages to pcap
- ASTERISK-28921: Wrong return value check for fwrite when writing to pcap
- file
- Reported by: nappsoft
- * [c16937cdbe] Pirmin Walthert -- res_pjsip_logger.c: correct the return
- value checks when writing to pcap
-
- Category: Resources/res_pjsip_messaging
-
- ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be
- changed
- Reported by: Alex
- * [03d24ca4c1] Sean Bright -- res_pjsip_messaging: Allow Content-Type to
- be overridden
- ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
- error when sending
- Reported by: Dmitriy Serov
- * [b1ca2c5d71] Sean Bright -- res_pjsip_messaging: Ensure
- MESSAGE_SEND_STATUS is set properly
-
- Category: Resources/res_pjsip_mwi
-
- ASTERISK-28575: MWI Send Notify Crash on 16.6
- Reported by: Joshua Elson
- * [5dae803eea] 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
- * [12dbeb69b0] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object
- for mwi containers
-
- Category: Resources/res_pjsip_nat
-
- ASTERISK-28884: x-ast-orig-host not filtered out from request URI and To
- header
- Reported by: nappsoft
- * [1399f8b4fe] Pirmin Walthert -- res_pjsip_nat.c: remove
- x-ast-orig-host from request URI and To header
-
- Category: Resources/res_pjsip_notify
-
- ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present
- instead of just one
- Reported by: AvayaXAsterisk
- * [90af050fa4] Sean Bright -- res_pjsip_notify: Only allow a single
- Event header to be added to a NOTIFY
-
- Category: Resources/res_pjsip_outbound_registration
-
- ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first
- entry in a SRV record set
- Reported by: George Joseph
- * [78b01f41ae] George Joseph -- res_pjsip_outbound_registration: Fix SRV
- failover on timeout
- ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
- Reported by: Kevin Harwell
- * [d5d41409e2] Kevin Harwell -- res_pjsip_outbound_registration: add
- support for SRV failover
- ASTERISK-28521: pjsip: Memory Leak
- Reported by: Mark
- * [cc83e76aa5] 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
- * [982a5025b3] Sean Bright -- res_pjsip_registrar: Validate Contact URI
- before adding to responses
-
- Category: Resources/res_pjsip_pubsub
-
- ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate
- (ASTERISK-27759) Causes Segfaults
- Reported by: Ross Beer
- * [a1f0c833ab] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence
- data ref when recreating.
- ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not
- preserve XML version number
- Reported by: Bryan Nelson
- * [4e7adbd8f4] Joshua C. Colp -- res_pjsip_pubsub: Add ability to
- persist generator state information.
-
- Category: Resources/res_pjsip_registrar
-
- ASTERISK-28995: res_pjsip_registrar: Expires on statically configured
- contacts is not correct
- Reported by: tootai
- * [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an
- expiration for static contacts.
-
- Category: Resources/res_pjsip_sdp_rtp
-
- ASTERISK-28784: res_pjsip_sdp_rtp: Only do hold/unhold on first audio
- stream
- Reported by: Joshua C. Colp
- * [34750d2068] Joshua C. Colp -- res_pjsip_sdp_rtp: Only do hold/unhold
- on default audio stream.
- ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during
- direct-media (native_rtp) bridge
- Reported by: Michael Neuhauser
- * [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore
- rtptimeout if direct-media is active
- ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold
- Reported by: Ross Beer
- * [77c9ba8e63] Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH
- transitions
- ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used
- Reported by: Torrey Searle
- * [bf4340f0ec] Torrey Searle -- res_pjsip_sdp_rtp: implement hold state
- handling on moh_passthrough
- 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
- * [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on
- created streams for incoming SDP.
-
- Category: Resources/res_pjsip_session
-
- ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due
- to codec negotiation after upgrading from Asterisk 16
- Reported by: Ross Beer
- * [5cc4a391b3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call
- pref.
- ASTERISK-29033: res_pjsip_session: Aggressively terminates session on
- failed re-INVITE
- Reported by: Joshua C. Colp
- * [3c074038fe] Joshua C. Colp -- res_pjsip_session: Don't aggressively
- terminate on failed re-INVITE.
- ASTERISK-28953: res_pjsip_session: Preserve stream label
- Reported by: Joshua C. Colp
- * [ee8ea9275f] Joshua C. Colp -- res_pjsip_session: Preserve label on
- incoming re-INVITE.
- ASTERISK-28871: res_pjsip_session: Unnecessary re-Invite on call answer
- Reported by: Alexei Gradinari
- * [afa2c9a868] Joshua C. Colp -- bridge: Don't try to match audio
- formats.
- ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have
- reflected state
- Reported by: Joshua C. Colp
- * [9620ecbf80] Joshua C. Colp -- res_pjsip_session: Don't restrict
- non-audio default streams to sendrecv.
- ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes
- Reported by: Joshua C. Colp
- * [ac155decae] Joshua C. Colp -- res_pjsip_session: Fix off-nominal
- session refreshes.
- 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
- * [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on
- created streams for incoming SDP.
- ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
- hangup when TEST_FRAMEWORK enabled
- Reported by: Bernhard Schmidt
- * [6ee1f1f507] 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
- * [c03f50c1c8] 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
- * [330ffa2bce] Salah Ahmed -- res_pjsip_t38: T.38 error correction mode
- selection at 200 ok received
-
- Category: Resources/res_realtime
-
- ASTERISK-21794: CLI command 'realtime update2' syntax failure when using
- according to usage help
- Reported by: Cedric BASSAGET
- * [094e87b0dc] Sean Bright -- res_realtime: Fix 'realtime update2'
- argument handling
-
- Category: Resources/res_rtp_asterisk
-
- ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string
- to each message block.
- Reported by: Thomas Johnson
- * [5ec7099312] Sean Bright -- bridge_channel: Ensure text messages are
- zero terminated
- ASTERISK-28939: res_rtp_asterisk: Don't have send/receive buffers on
- non-WebRTC
- Reported by: Joshua C. Colp
- * [c84d962eae] Joshua C. Colp -- res_rtp_asterisk: Don't assume setting
- retrans props means to enable.
- ASTERISK-28904: RTP ICE leaks the memory
- Reported by: sungtae kim
- * [c8c94b6cf1] sungtae kim -- res_rtp_asterisk.c: Fixed memory leak
- ASTERISK-28852: Unprotected access to nochecksums variable, causes build
- failures
- Reported by: Guido Falsi
- * [e4366308e1] Guido Falsi -- res_rtp_asterisk: Protect access to
- nochecksums with #ifdef
- ASTERISK-28827: res_rtp_asterisk: Loop when receive buffer is flushed by a
- received packet that is also in receive buffer with NACK
- Reported by: nappsoft
- * [d50fd0acc0] Pirmin Walthert -- res_rtp_asterisk: Resolve loop when
- receive buffer is flushed
- ASTERISK-28826: res_rtp_asterisk: Duplicate seqnos being added to send
- buffer with NACK
- Reported by: nappsoft
- * [ca032d1e2e] Pirmin Walthert -- res_rtp_asterisk: Free payload when
- error on insertion to data buffer
- ASTERISK-28812: First DTMF is not get
- Reported by: Bernard Merindol
- * [7db03e12a7] Bernard Merindol -- res_rtp_asterisk.c:Â Check for first
- DTMF having timestamp set to 0
- ASTERISK-28809: [patch] res_rtp_asterisk: Avoid absolute value on unsigned
- subtraction.
- Reported by: Alexander Traud
- * [1ef1b1b0c2] Alexander Traud -- res_rtp_asterisk: Avoid absolute value
- on unsigned subtraction.
- ASTERISK-28773: Incorrect Sender SSRC in RTCP when p2p rtp bridge is
- active
- Reported by: Torrey Searle
- * [a1dba820cf] Torrey Searle -- res_rtp_asterisk: Send correct sender
- SSRC when p2p bridge in use
- ASTERISK-28769: DTLS Handshake Fails to Occur if ice_support is enabled
- but not used
- Reported by: Torrey Searle
- * [14ba1806f3] Torrey Searle -- res_pjsip_sdp_rtp: Don't wait for ICE if
- not negotiated
- ASTERISK-28759: A non negotiated rtp frame causes call disconnection when
- there is a SSRC change
- Reported by: Paulo Vicentini
- * [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when
- remote SSRC changes.
- ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling
- Reported by: Joshua C. Colp
- * [87fda066ea] Joshua C. Colp -- res_rtp_asterisk: Improve video
- performance in certain networks.
- ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before
- allowing sending
- Reported by: Benjamin Keith Ford
- * [168637cc0c] Ben Ford -- RTP/ICE: Send on first valid pair.
- ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete
- dtls/srtp setup
- Reported by: Kevin Harwell
- * [3865b3fd6a] Kevin Harwell -- res_rtp_asterisk: bad audio (static) due
- to incomplete dtls/srtp setup
- ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet
- length doesn't match
- Reported by: Joshua Elson
- * [02129ad4d0] Joshua Colp -- res_rtp_asterisk: Always return provided
- DTLS packet length.
-
- Category: Resources/res_sorcery_memory_cache
-
- ASTERISK-28942: res_sorcery_memory_cache: Individual object expiration
- behaves unexpectedly with full backend caching
- Reported by: Joshua C. Colp
- * [a143c3a7b7] Joshua C. Colp -- res_sorcery_memory_cache: Disallow
- per-object expire with full backend.
-
- Category: Resources/res_speech
-
- ASTERISK-29040: res_speech: Assertion on format
- Reported by: Nickolay V. Shmyrev
- * [0319e0b07f] Nickolay Shmyrev -- res_speech: Bump reference on format
- object
-
- Category: Resources/res_srtp
-
- ASTERISK-28903: res_srtp: Answered Crypto Suite might be wrong in
- SDP/SDES.
- Reported by: Alexander Traud
- * [4de0e50c32] Alexander Traud -- res_srtp: Set all possible flags while
- selecting the Crypto Suite.
- ASTERISK-22920: Crash while Forwarding from TLS extension with CHANNEL
- args secure_bridge_media and secure_bridge_signaling
- Reported by: Shlomi Gutman
- * [29070b61f7] Alexander Traud -- core_local: Local calls are always
- secure.
-
- Category: Resources/res_stasis
-
- ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info
- Reported by: sungtae kim
- * [2e32b56bdb] sungtae kim -- stasis_bridge.c: Fixed wrong video_mode
- shown
- ASTERISK-28423: ARI causes STASIS Deadlock
- Reported by: Ross Beer
- * [cce2b0da95] Kevin Harwell -- stasis/app: don't lock an app before a
- call to send
- * [990a91b44a] George Joseph -- stasis: Don't hold app_registry and
- session locks unnecessarily
- ASTERISK-28633: stasis bridge topic leak
- Reported by: Joeran Vinzens
- * [1c9ddad4db] George Joseph -- stasis.c: Use correct topic name in
- stasis_topic_pool_delete_topic
-
- Category: Resources/res_stasis_playback
-
- ASTERISK-28713: res_stasis_playback: Error building JSON
- Reported by: Sébastien Duthil
- * [31dc904380] Sean Bright -- res_stasis_playback: Prevent media_index
- from going out of bounds
-
- Category: Resources/res_statsd
-
- ASTERISK-24484: Update documentation for statsd module - usage
- requirements unclear
- Reported by: Dan Jenkins
- * [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does
- nothing on its own
-
- Category: Tests/General
-
- ASTERISK-28808: [patch] test_stasis: Avoid always true warning with clang.
- Reported by: Alexander Traud
- * [bb28ed0d1b] Alexander Traud -- test_stasis: Avoid always true warning
- with clang.
-
- Category: Tests/testsuite
-
- ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming
- Language is python-2.7.
- Reported by: Alexander Traud
- * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API
- when possibly needed only.
- ASTERISK-28789: test_utils: incorrectly printing error 'declined to load'
- Reported by: Alexander Traud
- * [fc07eeaba1] Alexander Traud -- test_utils: Avoid incorrect error
- message on load.
-
- Category: Utilities/General
-
- ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling
- troubles
- Reported by: Sebastian Kemper
- * [b7fbb9c41f] Sebastian Kemper -- check_expr2: fix
- cross-compile/hardening issues
-
- Category: pjproject/pjsip
-
- ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered
- INVITE when NAT is active (UDP transport with external_media_address)
- Reported by: Michael Neuhauser
- * [6482ab5bea] Michael Neuhauser -- pjproject: clone sdp to protect
- against (nat) modifications
- ASTERISK-28929: pjproject_bundled: Honor --without-pjproject.
- Reported by: Alexander Traud
- * [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor
- --without-pjproject.
- ASTERISK-28794: res_pjsip: Crash when escaping during URI printing
- Reported by: nappsoft
- * [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing
- the contact_user value.
- ASTERISK-28859: pjsip: Increase maximum candidate count
- Reported by: Joshua C. Colp
- * [3078a00a6d] Joshua C. Colp -- pjsip: Increase maximum ICE candidate
- count.
- ASTERISK-28811: Crash occurs when fax session switches from T.38 to audio
- Reported by: Alexey Vasilyev
- * [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP
- re-negotiation scenarios.
- ASTERISK-28837: pjproject_bundled: Honor --without-pjproject.
- Reported by: Alexander Traud
- * [966acc6251] Alexander Traud -- pjproject_bundled: Honor
- --without-pjproject.
- ASTERISK-28758: pjsip startup errors when using "with-ssl" configure
- option
- Reported by: Patrick Wakano
- * [3431949a52] Alexander Traud -- pjproject_bundled: Repair ./configure
- --with-ssl without ARG.
- ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6
- Address Delimited by "[]" Rejected
- Reported by: Peter Sokolov
- * [9d9bde76a9] Sean Bright -- pjproject_bundled: Allow brackets in via
- parameters
- ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
- Reported by: Niklas Larsson
- * [5d9f9f4871] 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
- * [0844d6b127] Dan Cropp -- pjproject: Configurable setting for cnonce
- to include hyphens or not
-
- Improvement
-
- Category: Addons/chan_mobile
-
- ASTERISK-28832: chan_mobile creates PCMA streams that make some VoIP
- clients crash or not render received audio
- Reported by: Peter Turczak
- * [3303defd3f] Peter Turczak -- chan_mobile: Add smoother to make
- SIP/RTP endpoints happy.
-
- Category: Applications/NewFeature
-
- ASTERISK-28484: Add AudioSocket support
- Reported by: Seán C. McCord
- * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application,
- and ARI support.
-
- Category: Applications/app_confbridge
-
- ASTERISK-28658: app_confbridge: Add support for setting maximum sample
- rate
- Reported by: Joshua C. Colp
- * [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying
- maximum sample rate.
-
- Category: Applications/app_mixmonitor
-
- ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name
- Extension In MixMonitor
- Reported by: xrobau
- * [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to
- correct value when wav49 is used
-
- Category: Applications/app_page
-
- ASTERISK-27946: dial (API): Storage of dialed target uses
- AST_MAX_EXTENSION when it shouldn't
- Reported by: Joshua Elson
- * [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character
- limitation
-
- 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
- * [7362647e2f] 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
- * [2d67dbfef5] cmaj -- app_voicemail.c: Support multiple file formats
- for forwarded messages.
-
- Category: Bridges/bridge_native_rtp
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
-
- Category: Bridges/bridge_simple
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
-
- Category: Bridges/bridge_softmix
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
- ASTERISK-28658: app_confbridge: Add support for setting maximum sample
- rate
- Reported by: Joshua C. Colp
- * [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying
- maximum sample rate.
-
- Category: Channels/NewFeature
-
- ASTERISK-28484: Add AudioSocket support
- Reported by: Seán C. McCord
- * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application,
- and ARI support.
-
- Category: Channels/chan_pjsip
-
- ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
- Reported by: cmaj
- * [fe3cce816c] Richard Mudgett -- app_chanisavail.c: Simplify dialplan
- using ChanIsAvail.
- * [abcb4ab321] Richard Mudgett -- app_dial.c: Simplify dialplan using
- Dial.
- * [d86a6ac5ce] Richard Mudgett -- app_page.c: Simplify dialplan using
- Page.
-
- Category: Contrib/General
-
- ASTERISK-28726: install_prereq script uses the interactive mode when
- installing aptitude
- Reported by: Sylvain Afchain
- * [0c02d0a450] Sylvain Afchain -- install_prereq: Install aptitude
- non-interactively
-
- Category: Core/CodecInterface
-
- ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
- Reported by: Florian Floimair
- * [c18983207d] Florian Floimair -- core: Add H.265/HEVC passthrough
- support
-
- Category: Core/DNS
-
- ASTERISK-28853: Missing include on FreeBSD
- Reported by: Guido Falsi
- * [97494d8984] Guido Falsi -- core/dns: Add system include required on
- FreeBSD
-
- Category: Core/Dial
-
- ASTERISK-27946: dial (API): Storage of dialed target uses
- AST_MAX_EXTENSION when it shouldn't
- Reported by: Joshua Elson
- * [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character
- limitation
-
- Category: Core/HTTP
-
- ASTERISK-28750: TLS/SSL Key too small error
- Reported by: Martin Zeh
- * [7f2d56fc8c] Sean Bright -- tcptls.c: Log more informative OpenSSL
- errors
- ASTERISK-28710: Should be able to disable the /httpstatus URI in the
- built-in HTTP server
- Reported by: Sean Bright
- * [0dce6f746b] Sean Bright -- http: Add ability to disable /httpstatus
- URI
-
- Category: Core/ManagerInterface
-
- ASTERISK-28945: AMI SendText - add Content-Type parameter
- Reported by: Kevin Harwell
- * [cfed0ea033] Kevin Harwell -- manager - Add Content-Type parameter to
- the SendText action
-
- Category: Core/Streams
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
-
- Category: Documentation
-
- ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name
- Extension In MixMonitor
- Reported by: xrobau
- * [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to
- correct value when wav49 is used
- ASTERISK-28673: GET FULL VARIABLE documentation clarification
- Reported by: Jonathan Harris
- * [7d94bdde9d] Sean Bright -- res_agi: Improve GET FULL VARIABLE
- documentation
- ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
- Reported by: Sam Banks
- * [0dc7e29dd8] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling
- correetions.
-
- Category: Functions/func_volume
-
- ASTERISK-28813: func_volume: Allow decimal numbers as parameter to improve
- granularity
- Reported by: Jean Aunis - Prescom
- * [de66713fd5] Jean Aunis -- func_volume: Accept decimal number as
- argument
-
- Category: Resources/NewFeature
-
- ASTERISK-28484: Add AudioSocket support
- Reported by: Seán C. McCord
- * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application,
- and ARI support.
-
- Category: Resources/res_ari_bridges
-
- ASTERISK-28629: [patch] Add an "inhibitCOLP" flag to the bridges REST API
- Reported by: Jean Aunis - Prescom
- * [034ac357ad] Jean Aunis -- ARI: Ability to inhibit COLP frames when
- adding channels to a bridge
-
- Category: Resources/res_ari_channels
-
- ASTERISK-28896: ari: Add support for specifying variables on channel
- create
- Reported by: Joshua C. Colp
- * [15cbff9d54] Joshua C. Colp -- ari: Allow variables to be set on
- channel create.
-
- Category: Resources/res_http_websocket
-
- ASTERISK-28958: Continue reading string when ping received by websocket
- Reported by: Nickolay V. Shmyrev
- * [7163efd934] Nickolay Shmyrev -- res_http_websocket.c: Continue
- reading after ping/pong
- ASTERISK-28949: res_http_websocket: Add masking to websocket client
- Reported by: Moises Silva
- * [9445dac43b] Moises Silva -- res_http_websocket: Add payload masking
- to the websocket client
-
- Category: Resources/res_pjsip
-
- ASTERISK-28959: res_pjsip: Added option for disable rport parameter set
- Reported by: sungtae kim
- * [81b5e4a73f] sungtae kim -- res_pjsip.c: Added disable_rport option
- for pjsip.conf
- ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
- Reported by: Kevin Harwell
- * [2ee455958e] George Joseph -- codec_negotiation: Implement
- outgoing_call_offer_pref
- ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
- Reported by: Kevin Harwell
- * [06dada3f01] Kevin Harwell -- codec negotiation: add
- incoming_call_offer_prefs option
- ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold
- re-invites
- Reported by: Torrey Searle
- * [b43cdc7f1e] Torrey Searle -- channel/chan_pjsip: add dialplan
- function for music on hold
-
- Category: Resources/res_pjsip_logger
-
- ASTERISK-28895: res_pjsip_logger: Add tons'o'functionality
- Reported by: Joshua C. Colp
- * [a7aaee70c6] Joshua C. Colp -- res_pjsip_logger: Expand functionality
- to improve logging.
-
- Category: Resources/res_pjsip_outbound_registration
-
- ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
- Reported by: Daniel
- * [e73eba85c1] Joshua Colp -- res_pjsip_outbound_registration: Extend
- documentation for "max_retries".
-
- Category: Resources/res_pjsip_sdp_rtp
-
- ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
- Reported by: Kevin Harwell
- * [2ee455958e] George Joseph -- codec_negotiation: Implement
- outgoing_call_offer_pref
- ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
- Reported by: Kevin Harwell
- * [06dada3f01] Kevin Harwell -- codec negotiation: add
- incoming_call_offer_prefs option
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
-
- Category: Resources/res_pjsip_session
-
- ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
- Reported by: Kevin Harwell
- * [2ee455958e] George Joseph -- codec_negotiation: Implement
- outgoing_call_offer_pref
- ASTERISK-28782: Add support for Content-Disposition header in multi-part
- INVITES
- Reported by: Torrey Searle
- * [e12244153a] Torrey Searle -- res_pjsip_session: implement processing
- of Content-Disposition
- ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add
- video
- Reported by: Joshua C. Colp
- * [21e9051461] Joshua C. Colp -- res_pjsip_session: Apply intention
- behind requested formats.
- ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
- Reported by: Kevin Harwell
- * [06dada3f01] Kevin Harwell -- codec negotiation: add
- incoming_call_offer_prefs option
-
- Category: Third-Party/pjproject
-
- ASTERISK-28866: third-party/pjproject/configure.m4 contains bashisms
- Reported by: Guido Falsi
- * [c831f03273] Guido Falsi -- pjproject: Remove bashism from
- configure.m4 script
-
- Category: pjproject/pjsip
-
- ASTERISK-28899: Upgrade Asterisk to bundled pjproject 2.10
- Reported by: Kevin Harwell
- * [415b55af5a] Kevin Harwell -- pjproject: Upgrade bundled version to
- pjproject 2.10
- ASTERISK-28879: pjproject has race conditions in it's build system
- Reported by: Guido Falsi
- * [801d570f6e] Guido Falsi -- pjproject: Fix race condition when
- building with parallel make
-
- ----------------------------------------------------------------------
-
- 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: Channels/chan_sip/TCP-TLS
-
- ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
- Reported by: Alexander Traud
- * [da9554d925] Alexander Traud -- chan_sip: TCP/TLS client without
- server.
-
- Category: Core/Configuration
-
- ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues
- Reported by: EDV O-TON
- * [eb9252ea27] Sean Bright -- res_config_odbc: Preserve empty strings
- returned by the database
-
- ----------------------------------------------------------------------
-
- 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 |
- |------------+-------------------+---------------------------------------|
- | 6fd94258f8 | Asterisk | Update for 18.0.0-rc2 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | 704cb88799 | Asterisk | Update for 18.0.0-rc1 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | f589985840 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 18.0.0 |
- |------------+-------------------+---------------------------------------|
- | | | res_pjsip_session: Deferred re-INVITE |
- | 5a49757e40 | Patrick Verzele | without SDP send a=sendrecv instead |
- | | | of a=sendonly |
- |------------+-------------------+---------------------------------------|
- | ec03909831 | Kevin Harwell | conversions: Add string to signed |
- | | | integer conversion functions |
- |------------+-------------------+---------------------------------------|
- | e32815dddb | George Joseph | ast_coredumper: Fix issues with |
- | | | naming |
- |------------+-------------------+---------------------------------------|
- | 9ed1b1452d | Alexander Traud | sip_nat_settings: Update script for |
- | | | latest Linux. |
- |------------+-------------------+---------------------------------------|
- | 217449a1e5 | Alexander Traud | samples: Fix keep_alive_interval |
- | | | default in pjsip.conf. |
- |------------+-------------------+---------------------------------------|
- | 5a8cacb93d | George Joseph | logger.c: Added a new log formatter |
- | | | called "plain" |
- |------------+-------------------+---------------------------------------|
- | | | res_musiconhold.c: Use |
- | 5dfeeba623 | Sean Bright | ast_file_read_dir to scan MoH |
- | | | directory |
- |------------+-------------------+---------------------------------------|
- | c4c72d55a2 | George Joseph | scope_trace: Added debug messages and |
- | | | added additional macros |
- |------------+-------------------+---------------------------------------|
- | d26ab7f8f9 | George Joseph | stream.c: Added 2 more debugging |
- | | | utils and added pos to stream string |
- |------------+-------------------+---------------------------------------|
- | 6faf76308d | George Joseph | ACN: Changes specific to the core |
- |------------+-------------------+---------------------------------------|
- | a3d87f78ed | Joshua C. Colp | res_pjsip: Fix codec preference |
- | | | defaults. |
- |------------+-------------------+---------------------------------------|
- | | | vector.h: Fix implementation of |
- | da8a617dc9 | Sean Bright | AST_VECTOR_COMPACT() for empty |
- | | | vectors |
- |------------+-------------------+---------------------------------------|
- | 769a9611e7 | Ben Ford | utils.c: NULL terminate |
- | | | ast_base64decode_string. |
- |------------+-------------------+---------------------------------------|
- | 802aa97fa0 | George Joseph | ACN: Configuration renaming for pjsip |
- | | | endpoint |
- |------------+-------------------+---------------------------------------|
- | de23cb4002 | Ben Ford | res_stir_shaken: Fix memory |
- | | | allocation error in curl.c |
- |------------+-------------------+---------------------------------------|
- | 71446b68fc | George Joseph | res_pjsip_session: Ensure reused |
- | | | streams have correct bundle group |
- |------------+-------------------+---------------------------------------|
- | d9ae902f52 | Sean Bright | utf8.c: Add UTF-8 validation and |
- | | | utility functions |
- |------------+-------------------+---------------------------------------|
- | 9022f35f09 | Sean Bright | vector.h: Add AST_VECTOR_SORT() |
- |------------+-------------------+---------------------------------------|
- | a678dafac8 | George Joseph | CI: Force publishAsteriskDocs to use |
- | | | python2 |
- |------------+-------------------+---------------------------------------|
- | af70bbb13a | Joshua C. Colp | websocket / pjsip: Increase maximum |
- | | | packet size. |
- |------------+-------------------+---------------------------------------|
- | 8d15f72721 | Joshua C. Colp | pjsip: Include timer patch to prevent |
- | | | cancelling timer 0. |
- |------------+-------------------+---------------------------------------|
- | 3330764213 | George Joseph | Update .gitreview defaultbranch to 18 |
- |------------+-------------------+---------------------------------------|
- | 1f5e6805bf | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 18.0.0 |
- |------------+-------------------+---------------------------------------|
- | 5fbed5af24 | Ben Ford | res_stir_shaken: Add stir_shaken |
- | | | option and general improvements. |
- |------------+-------------------+---------------------------------------|
- | e88beedd08 | George Joseph | res_pjsip_session: Fix segv in |
- | | | session_on_rx_response |
- |------------+-------------------+---------------------------------------|
- | 9bd1d686a1 | George Joseph | ACN: Add tracing to existing code |
- |------------+-------------------+---------------------------------------|
- | 2d22e34206 | George Joseph | ACN: res_pjsip endpoint options |
- |------------+-------------------+---------------------------------------|
- | d093e44b1e | George Joseph | frame.c: Make debugging easier |
- |------------+-------------------+---------------------------------------|
- | 955b7b4fdb | George Joseph | Scope Trace: Make it easier to trace |
- | | | through synchronous tasks |
- |------------+-------------------+---------------------------------------|
- | 8d1064eaaf | George Joseph | Streams: Add features for Advanced |
- | | | Codec Negotiation |
- |------------+-------------------+---------------------------------------|
- | 7440fd0397 | George Joseph | Scope Trace: Add some new tracing |
- | | | macros and an ast_str helper |
- |------------+-------------------+---------------------------------------|
- | 1274117102 | Ben Ford | res_stir_shaken: Add outbound INVITE |
- | | | support. |
- |------------+-------------------+---------------------------------------|
- | | | res_pjsip: Include <pjsip_ua.h> |
- | f1cfd54976 | Walter Doekes | instead of internal |
- | | | "pjsua-lib/pjsua.h" |
- |------------+-------------------+---------------------------------------|
- | b9f42a717e | George Joseph | app_confbridge: Plug ref leak of |
- | | | bridge channel with send_events |
- |------------+-------------------+---------------------------------------|
- | 3d1bf3c537 | Kevin Harwell | Compiler fixes for gcc 10 |
- |------------+-------------------+---------------------------------------|
- | 559fa0e89c | Ben Ford | cli.c: Fix compiler error. |
- |------------+-------------------+---------------------------------------|
- | 3927f79cb5 | Ben Ford | res_stir_shaken: Add inbound INVITE |
- | | | support. |
- |------------+-------------------+---------------------------------------|
- | 1fcb6b1b21 | Joshua C. Colp | bridge_channel: Don't queue unmapped |
- | | | frames. |
- |------------+-------------------+---------------------------------------|
- | ca3c22c5f1 | George Joseph | Scope Tracing: A new facility for |
- | | | tracing scope enter/exit |
- |------------+-------------------+---------------------------------------|
- | ec7890d7c6 | Joshua C. Colp | res_sorcery_config: Always reload |
- | | | configuration on errors. |
- |------------+-------------------+---------------------------------------|
- | f506cc4896 | Ben Ford | res_stir_shaken: Add unit tests for |
- | | | signing and verification. |
- |------------+-------------------+---------------------------------------|
- | e29df34de0 | Ben Ford | res_stir_shaken: Added dialplan |
- | | | function and API call. |
- |------------+-------------------+---------------------------------------|
- | 44e5dd288b | Jaco Kroon | Remove #include <sys/cdefs.h> |
- |------------+-------------------+---------------------------------------|
- | 1cfd30bd8a | Joshua C. Colp | res_stir_shaken: Use ast_asprintf for |
- | | | creating file path. |
- |------------+-------------------+---------------------------------------|
- | 9acf840f7c | Ben Ford | res_stir_shaken: Implemented |
- | | | signature verification. |
- |------------+-------------------+---------------------------------------|
- | 7baf2c4bf1 | George Joseph | app_voicemail: Add workaround for a |
- | | | gcc 10 issue with -Wrestrict |
- |------------+-------------------+---------------------------------------|
- | 4ef5ba58f5 | Alexander Traud | BuildSystem: Only if found LibPRI, |
- | | | check its optional parts. |
- |------------+-------------------+---------------------------------------|
- | ef580f96e7 | Alexander Traud | BuildSystem: Only if found external |
- | | | PJProject, check its optional parts. |
- |------------+-------------------+---------------------------------------|
- | 611529fa52 | Alexander Traud | res_stir_shaken: Do not build without |
- | | | OpenSSL. |
- |------------+-------------------+---------------------------------------|
- | | | res_audiosocket: Avoid |
- | 27de0c9700 | Alexander Traud | Sometimes-uninitialized Warning with |
- | | | Clang. |
- |------------+-------------------+---------------------------------------|
- | 2b80e5f5da | Jaco Kroon | res_rtp_asterisk: iterate all local |
- | | | addresses looking to populate ICE. |
- |------------+-------------------+---------------------------------------|
- | 1cf569ba2b | Jaco Kroon | res_pjsip: document legal dtls_verify |
- | | | endpoint options. |
- |------------+-------------------+---------------------------------------|
- | 52ecbbd014 | Alexander Traud | _pjsua: Build even with Clang. |
- |------------+-------------------+---------------------------------------|
- | ee1c7f465b | Alexander Traud | res_rtp_asterisk: Build without |
- | | | PJProject. |
- |------------+-------------------+---------------------------------------|
- | | | Revert "res_config_odbc: Preserve |
- | 60925c68e8 | Sean Bright | empty strings returned by the |
- | | | database" |
- |------------+-------------------+---------------------------------------|
- | c5f3836bcc | Jaco Kroon | main/backtrace: binutils-2.34 fix. |
- |------------+-------------------+---------------------------------------|
- | 7ba6d43083 | George Joseph | test_res_pjsip_session_caps: Create |
- | | | unit test |
- |------------+-------------------+---------------------------------------|
- | 57a457c26c | Ben Ford | res_stir_shaken: Implemented signing |
- | | | of JSON payload. |
- |------------+-------------------+---------------------------------------|
- | d32e559e8a | Jaco Kroon | acl: implement a centralized ACL |
- | | | output mechanism for HAs and ACLs. |
- |------------+-------------------+---------------------------------------|
- | 1b6c58896f | Joshua C. Colp | chan_sip: Send 403 when ACL fails. |
- |------------+-------------------+---------------------------------------|
- | 3ed80fc57b | Joshua C. Colp | CHANGES: Change md file extension to |
- | | | txt. |
- |------------+-------------------+---------------------------------------|
- | 26713dc88b | Kevin Harwell | ast_coredumper: add Asterisk |
- | | | information dump |
- |------------+-------------------+---------------------------------------|
- | 6f731f153b | Jaco Kroon | netsock2: compile fixes. |
- |------------+-------------------+---------------------------------------|
- | 211bb8a79c | Ben Ford | res_stir_shaken: Initial commit and |
- | | | reading private key. |
- |------------+-------------------+---------------------------------------|
- | a699e016dd | Jaco Kroon | build: enable building with uClibc |
- |------------+-------------------+---------------------------------------|
- | f824cd6a13 | Jaco Kroon | build: search from newest to oldest |
- | | | for gmime. |
- |------------+-------------------+---------------------------------------|
- | 82c3939c38 | Jaco Kroon | res_rtp_asterisk: implement ACL |
- | | | mechanism for ICE and STUN addresses. |
- |------------+-------------------+---------------------------------------|
- | 2ad64e97c0 | Jaco Kroon | Update main/backtrace.c to deal with |
- | | | changes in binutils 2.34. |
- |------------+-------------------+---------------------------------------|
- | 49cf84578e | Sean Bright | chan_vpb: Fix 'catching polymorphic |
- | | | type ... by value' error |
- |------------+-------------------+---------------------------------------|
- | d68f940f6e | Sean Bright | dns_txt: Add TXT record parsing |
- | | | support |
- |------------+-------------------+---------------------------------------|
- | 00a7e4b51d | George Joseph | CI: Create generic jenkinsfile |
- |------------+-------------------+---------------------------------------|
- | e089779908 | Rodrigo RamÃrez | res_rtp_asterisk: Add 'rtp show |
- | | Norambuena | settings' cli command |
- |------------+-------------------+---------------------------------------|
- | 680e6b9774 | Walter Doekes | app_queue: Refactor odd placement of |
- | | | if's around say_position |
- |------------+-------------------+---------------------------------------|
- | 1e1651b4f4 | Kevin Harwell | format_cap: make function parameters |
- | | | 'const' |
- |------------+-------------------+---------------------------------------|
- | 0b5c6fddf1 | Walter Doekes | say: Remove unused "plural" option |
- | | | from main/say |
- |------------+-------------------+---------------------------------------|
- | 5cd7230f3c | Jaco Kroon | addons/res_config_mysql: silense |
- | | | warnings about printf format errors. |
- |------------+-------------------+---------------------------------------|
- | de6919f339 | Sean Bright | ast_tls_cert: Allow private key size |
- | | | to be set on command line |
- |------------+-------------------+---------------------------------------|
- | 8dcdce42a9 | Sean Bright | app_mixmonitor: Turn on |
- | | | synchronization by default |
- |------------+-------------------+---------------------------------------|
- | 0f6ee98c3f | Joshua C. Colp | stasis: Use format specifier for |
- | | | size_t. |
- |------------+-------------------+---------------------------------------|
- | 1e037ebb97 | Sean Bright | func_odbc: Prevent snprintf() |
- | | | truncation warning |
- |------------+-------------------+---------------------------------------|
- | a72caa041f | George Joseph | doc: Fix CHANGES entries to have .txt |
- | | | suffix and update READMEs |
- |------------+-------------------+---------------------------------------|
- | 1b53d329ac | Joshua C. Colp | res_rtp_asterisk: Don't produce |
- | | | transport-cc if no packets. |
- |------------+-------------------+---------------------------------------|
- | b76ab5e5c9 | George Joseph | message.c: Add option to suppress the |
- | | | Message channel AMI and ARI events |
- |------------+-------------------+---------------------------------------|
- | 113d05e504 | Walter Doekes | chan_sip: Clarify in sample docs how |
- | | | directmediapermit/-acl should be used |
- |------------+-------------------+---------------------------------------|
- | 262221f4d9 | Sean Bright | func_odbc.conf.sample: Add example |
- | | | lookup |
- |------------+-------------------+---------------------------------------|
- | f09cf4da44 | Sean Bright | app_voicemail: Remove MessageExists |
- | | | and MESSAGE_EXISTS() |
- |------------+-------------------+---------------------------------------|
- | 5cbf47714a | Sean Bright | app_voicemail, say: Fix various |
- | | | leading whitespace problems |
- |------------+-------------------+---------------------------------------|
- | | | netsock2: |
- | 3bc8b36537 | Jaco Kroon | ast_addressfamily_to_sockaddrsize and |
- | | | ast_sockaddr_from_sockaddr. |
- |------------+-------------------+---------------------------------------|
- | 00a7432156 | Kevin Harwell | app_agent_pool: Update XML docs for |
- | | | AgentLogin |
- |------------+-------------------+---------------------------------------|
- | 19069f7db7 | Richard Mudgett | app_bridgeaddchan.c: Make BridgeAdd |
- | | | be more like Bridge |
- |------------+-------------------+---------------------------------------|
- | 0376f2bba9 | Richard Mudgett | features.c: Make Bridge application |
- | | | tolerate unspecified channel. |
- |------------+-------------------+---------------------------------------|
- | 0d1f3d9bf3 | Richard Mudgett | app_chanspy.c: Reduce log message |
- | | | level from notice to verbose. |
- |------------+-------------------+---------------------------------------|
- | a457947198 | Richard Mudgett | app_softhangup.c: Reduce unnecessary |
- | | | warning to verbose message. |
- |------------+-------------------+---------------------------------------|
- | fc99ac8c9a | Sean Bright | db: Initialize condition primitive |
- | | | before use |
- |------------+-------------------+---------------------------------------|
- | 32160cb456 | Jaco Kroon | ACL: ast_apply_acl_nolog - identical |
- | | | to ast_apply_acl but without logging. |
- |------------+-------------------+---------------------------------------|
- | d0b198b330 | Joshua Colp | Revert "PJSIP_CONTACT: add missing |
- | | | argument documentation" |
- |------------+-------------------+---------------------------------------|
- | | | res_pjsip_registrar.c: Prevent |
- | 0183e2bc67 | Sean Bright | possible buffer overflow with domain |
- | | | aliases |
- |------------+-------------------+---------------------------------------|
- | fd823225a6 | Thomas Arimont | channel.c: Resolve issue with |
- | | | receiving SIP INFO packets for DTMF |
- |------------+-------------------+---------------------------------------|
- | 366da90f74 | George Joseph | CI: Turn off shallow cloning |
- | | | altogether |
- |------------+-------------------+---------------------------------------|
- | 91c3b5b09d | Sean Bright | media_cache.c: Various CLI |
- | | | improvements |
- |------------+-------------------+---------------------------------------|
- | 48161dfc71 | Rodrigo RamÃrez | queue_log: Add alembic script for |
- | | Norambuena | generate db table for queue_log |
- |------------+-------------------+---------------------------------------|
- | 2a6a2800e7 | George Joseph | CI: Fix missing script block in |
- | | | jenkinsfiles |
- |------------+-------------------+---------------------------------------|
- | 4abb54b2e4 | George Joseph | CI: Fix missing script block in |
- | | | jenkinsfiles |
- |------------+-------------------+---------------------------------------|
- | e8e1314fcb | George Joseph | CI: Increase clone depth and do |
- | | | better cleanup |
- |------------+-------------------+---------------------------------------|
- | a5fa0d662e | Sean Bright | res_pjsip_registrar: Fix uninitlized |
- | | | variable warning |
- |------------+-------------------+---------------------------------------|
- | f2d5ed54ea | Alexei Gradinari | serializer: set high/low alert levels |
- | | | on whole pool |
- |------------+-------------------+---------------------------------------|
- | bdd785d31c | Kevin Harwell | various files - fix some alerts |
- | | | raised by lgtm code analysis |
- |------------+-------------------+---------------------------------------|
- | 0e3b397812 | Kevin Harwell | res_pjsip_session: initialize |
- | | | pending's topology to endpoint's |
- |------------+-------------------+---------------------------------------|
- | 8a1f30af04 | Corey Farrell | core: Improve MALLOC_DEBUG for |
- | | | frames. |
- |------------+-------------------+---------------------------------------|
- | d71d0f9489 | George Joseph | ExternalMedia: Change return object |
- | | | from ExternalMedia to Channel |
- |------------+-------------------+---------------------------------------|
- | 6e907ae5d4 | Joshua Colp | res_rtp_asterisk: Remove a log |
- | | | message that slipped in. |
- |------------+-------------------+---------------------------------------|
- | a60d2e905c | Joshua Colp | test_res_rtp: Enable FIR and REMB |
- | | | nominal tests. |
- |------------+-------------------+---------------------------------------|
- | b27a5183da | Chris Savinovich | test_taskprocessor.c: Fix test |
- | | | failure on Ubuntu |
- |------------+-------------------+---------------------------------------|
- | c0efe19cec | Kevin Harwell | serializer: move/add asterisk |
- | | | serializer pool functionality |
- |------------+-------------------+---------------------------------------|
- | 2970a13fb8 | Kevin Harwell | res_pjsip/res_pjsip_mwi: use |
- | | | centralized serializer pools |
- |------------+-------------------+---------------------------------------|
- | 068ed2c626 | Alexei Gradinari | res_pjsip_pubsub: add endpoint to |
- | | | some warning |
- |------------+-------------------+---------------------------------------|
- | ba64d68273 | Jonathan Rose | basic-pbx: Bring forward queue |
- | | | configuration from 13 |
- |------------+-------------------+---------------------------------------|
- | 4c3655ecfd | Ben Ford | taskprocessor.c: Added "like" support |
- | | | to 'core show taskprocessors' |
- |------------+-------------------+---------------------------------------|
- | 966488ab52 | Sean Bright | res_musiconhold: Add new 'playlist' |
- | | | mode |
- |------------+-------------------+---------------------------------------|
- | f7045cefd9 | Corey Farrell | stasis_state: Create internal |
- | | | stasis_state_proxy object. |
- |------------+-------------------+---------------------------------------|
- | 67ba62f4e6 | Kevin Harwell | res_pjsip_pubsub: change warning to |
- | | | debug |
- |------------+-------------------+---------------------------------------|
- | 4de1e6d0e6 | Ben Ford | taskprocessor.c: Add CLI commands to |
- | | | reset taskprocessor stats. |
- |------------+-------------------+---------------------------------------|
- | 725e991faf | Corey Farrell | core: Add AO2_ALLOC_OPT_NO_REF_DEBUG |
- | | | option. |
- |------------+-------------------+---------------------------------------|
- | e82f2f6e82 | George Joseph | astmm.c: Display backtrace with |
- | | | memory show allocations |
- |------------+-------------------+---------------------------------------|
- | a4142c8437 | Corey Farrell | core: Fix ABI mismatch of |
- | | | ao2_global_obj. |
- |------------+-------------------+---------------------------------------|
- | ca608d2575 | Corey Farrell | stasis: refcounter.py can incorrectly |
- | | | report skewed objects. |
- |------------+-------------------+---------------------------------------|
- | 3dfbc05c53 | Corey Farrell | stasis: Fix leaks |
- |------------+-------------------+---------------------------------------|
- | 863fe2225f | Corey Farrell | app_voicemail: Fix module unload |
- | | | leak. |
- |------------+-------------------+---------------------------------------|
- | 723b695ce5 | Ben Ford | res_rtp_asterisk.c: Send RTCP as |
- | | | compound packets. |
- |------------+-------------------+---------------------------------------|
- | 0e56643d9f | Ben Ford | res_rtp: Add unit tests for RTCP |
- | | | stats. |
- |------------+-------------------+---------------------------------------|
- | 2ae1a22e0e | George Joseph | ARI: External Media |
- |------------+-------------------+---------------------------------------|
- | 5fb9b23105 | George Joseph | chan_sip: Update links referenced in |
- | | | deprecation notice |
- |------------+-------------------+---------------------------------------|
- | | | test_utils.c: Skip test |
- | ed757cc7bb | Chris-Savinovich | adsi_loaded_test if module not |
- | | | loaded. |
- |------------+-------------------+---------------------------------------|
- | 1d06a1efb3 | Igor Goncharovsky | chan_unistim: Fix code, causing all |
- | | | incoming DTMF sent back to asterisk |
- |------------+-------------------+---------------------------------------|
- | 649003821e | Igor Goncharovsky | chan_unistim: Fix RTP port byte order |
- | | | for big-endian arch |
- |------------+-------------------+---------------------------------------|
- | 3ef52b0b17 | Alexei Gradinari | Fix misname 'res_external_mwi' to |
- | | | 'res_mwi_external' in comments. |
- |------------+-------------------+---------------------------------------|
- | 19045db392 | George Joseph | chan_rtp: Accept hostname as well as |
- | | | ip address as destination |
- |------------+-------------------+---------------------------------------|
- | 9e015713cc | George Joseph | dns_core: Create new API |
- | | | ast_dns_resolve_ipv6_and_ipv4 |
- |------------+-------------------+---------------------------------------|
- | 8da4e28a81 | George Joseph | res_ari.c: Prefer exact handler match |
- | | | over wildcard |
- |------------+-------------------+---------------------------------------|
- | 64906c4c9b | Sean Bright | audiohook.c: Substitute silence for |
- | | | unavailable audio frames |
- |------------+-------------------+---------------------------------------|
- | 446bac733d | George Joseph | CI: Escape backslashes in |
- | | | printenv/sort/tr |
- |------------+-------------------+---------------------------------------|
- | be6130607d | George Joseph | CI: Add "throttle" label and |
- | | | "skip_gate" capability |
- |------------+-------------------+---------------------------------------|
- | c01dd2a41a | George Joseph | CI: Make node labels job-specific |
- |------------+-------------------+---------------------------------------|
- | 9d07d5a6d6 | Sean Bright | app_voicemail: Remove extra |
- | | | menuselect build options |
- |------------+-------------------+---------------------------------------|
- | 1f8ae708a0 | Sean Bright | res_musiconhold: Use a vector instead |
- | | | of custom array allocation |
- |------------+-------------------+---------------------------------------|
- | 5f66fb5139 | Sean Bright | manager: Send fewer packets |
- |------------+-------------------+---------------------------------------|
- | 5e6e1175d5 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 17.0.0 |
- |------------+-------------------+---------------------------------------|
- | 8d10028b98 | George Joseph | Update master for Asterisk 18 |
- |------------+-------------------+---------------------------------------|
- | 7ce9ee7f2e | Sean Bright | res_musiconhold: Use |
- | | | ast_pipe_nonblock() wrapper |
- |------------+-------------------+---------------------------------------|
- | 8e44d823c1 | 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.
-
- asterisk-17.0.0-summary.html | 1265 --
- asterisk-17.0.0-summary.txt | 2973 ----
- b/.gitreview | 2
- b/.version | 2
- b/CHANGES | 400
- b/ChangeLog | 6317 +++++++++-
- b/Makefile | 8
- b/Makefile.rules | 19
- b/README-SERIOUSLY.bestpractices.md | 4
- b/UPGRADE.txt | 120
- b/addons/cdr_mysql.c | 18
- b/addons/chan_mobile.c | 63
- b/addons/chan_ooh323.c | 7
- b/addons/ooh323c/src/decode.c | 10
- b/addons/ooh323c/src/ooSocket.c | 2
- b/addons/ooh323c/src/oochannels.c | 2
- b/addons/res_config_mysql.c | 16
- b/apps/app_agent_pool.c | 4
- b/apps/app_amd.c | 24
- b/apps/app_audiosocket.c | 240
- b/apps/app_bridgeaddchan.c | 62
- b/apps/app_cdr.c | 8
- b/apps/app_chanisavail.c | 137
- b/apps/app_chanspy.c | 3
- b/apps/app_confbridge.c | 9
- b/apps/app_dahdiras.c | 6
- b/apps/app_dial.c | 108
- b/apps/app_dictate.c | 4
- b/apps/app_fax.c | 2
- b/apps/app_followme.c | 10
- b/apps/app_getcpeid.c | 1
- b/apps/app_meetme.c | 2
- b/apps/app_minivm.c | 3
- b/apps/app_mixmonitor.c | 51
- b/apps/app_osplookup.c | 13
- b/apps/app_page.c | 30
- b/apps/app_playback.c | 9
- b/apps/app_queue.c | 201
- b/apps/app_readexten.c | 3
- b/apps/app_record.c | 3
- b/apps/app_senddtmf.c | 13
- b/apps/app_softhangup.c | 2
- b/apps/app_stack.c | 11
- b/apps/app_stream_echo.c | 2
- b/apps/app_voicemail.c | 839 -
- b/apps/confbridge/conf_config_parser.c | 30
- b/apps/confbridge/confbridge_manager.c | 2
- b/apps/confbridge/include/confbridge.h | 2
- b/asterisk-18.0.0-rc2-summary.html | 15
- b/asterisk-18.0.0-rc2-summary.txt | 83
- b/bridges/bridge_native_rtp.c | 173
- b/bridges/bridge_simple.c | 200
- b/bridges/bridge_softmix.c | 281
- b/cdr/cdr_odbc.c | 2
- b/cdr/cdr_pgsql.c | 2
- b/cel/cel_pgsql.c | 2
- b/channels/Makefile | 2
- b/channels/chan_audiosocket.c | 302
- b/channels/chan_dahdi.c | 39
- b/channels/chan_dahdi.h | 18
- b/channels/chan_iax2.c | 16
- b/channels/chan_motif.c | 9
- b/channels/chan_pjsip.c | 314
- b/channels/chan_rtp.c | 19
- b/channels/chan_sip.c | 268
- b/channels/chan_unistim.c | 178
- b/channels/chan_vpb.cc | 2
- b/channels/iax2/parser.c | 18
- b/channels/pjsip/cli_commands.c | 13
- b/channels/pjsip/dialplan_functions.c | 84
- b/channels/pjsip/include/dialplan_functions.h | 25
- b/channels/sig_pri.c | 25
- b/channels/sip/include/sip.h | 1
- 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/asterisk.conf.sample | 6
- b/configs/samples/confbridge.conf.sample | 7
- b/configs/samples/extconfig.conf.sample | 1
- b/configs/samples/func_odbc.conf.sample | 19
- b/configs/samples/http.conf.sample | 10
- b/configs/samples/indications.conf.sample | 11
- b/configs/samples/logger.conf.sample | 15
- b/configs/samples/musiconhold.conf.sample | 23
- b/configs/samples/pjsip.conf.sample | 79
- b/configs/samples/queues.conf.sample | 2
- b/configs/samples/rtp.conf.sample | 30
- b/configs/samples/sip.conf.sample | 4
- b/configs/samples/stir_shaken.conf.sample | 61
- b/configure | 666 -
- b/configure.ac | 196
- b/contrib/ast-db-manage/README.md | 1
- b/contrib/ast-db-manage/config/versions/61797b9fced6_add_stir_shaken.py | 31
- b/contrib/ast-db-manage/config/versions/79290b511e4b_pjsip_add_disable_rport.py | 39
- b/contrib/ast-db-manage/config/versions/b80485ff4dd0_add_pjsip_endpoint_acn_options.py | 29
- b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py | 54
- b/contrib/ast-db-manage/queue_log.ini.sample | 58
- b/contrib/ast-db-manage/queue_log/env.py | 1
- b/contrib/ast-db-manage/queue_log/script.py.mako | 24
- b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py | 38
- b/contrib/realtime/mysql/mysql_config.sql | 39
- b/contrib/realtime/postgresql/postgresql_config.sql | 45
- b/contrib/scripts/ast_coredumper | 421
- b/contrib/scripts/ast_tls_cert | 8
- b/contrib/scripts/install_prereq | 30
- b/contrib/scripts/sip_nat_settings | 19
- b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
- b/contrib/valgrind.supp | 14
- b/doc/CHANGES-staging/README.md | 8
- b/doc/CHANGES-staging/hide_messaging_ami_events | 11
- b/doc/UPGRADE-staging/README.md | 7
- b/doc/appdocsxml.dtd | 2
- b/formats/format_g726.c | 16
- b/formats/msgsm.h | 4
- b/funcs/func_aes.c | 4
- b/funcs/func_channel.c | 8
- b/funcs/func_curl.c | 49
- b/funcs/func_env.c | 5
- b/funcs/func_jitterbuffer.c | 19
- b/funcs/func_odbc.c | 26
- b/funcs/func_pjsip_contact.c | 6
- b/funcs/func_volume.c | 12
- b/include/asterisk/abstract_jb.h | 4
- b/include/asterisk/acl.h | 69
- b/include/asterisk/app.h | 7
- b/include/asterisk/ari.h | 2
- b/include/asterisk/astobj2.h | 61
- b/include/asterisk/audiohook.h | 2
- b/include/asterisk/autoconfig.h.in | 9
- b/include/asterisk/bridge.h | 21
- b/include/asterisk/bridge_features.h | 4
- b/include/asterisk/calendar.h | 4
- b/include/asterisk/channel.h | 87
- b/include/asterisk/channel_internal.h | 5
- b/include/asterisk/config.h | 18
- b/include/asterisk/config_options.h | 2
- b/include/asterisk/conversions.h | 54
- b/include/asterisk/core_unreal.h | 19
- b/include/asterisk/dns_core.h | 22
- b/include/asterisk/dns_internal.h | 31
- b/include/asterisk/dns_txt.h | 64
- b/include/asterisk/format_cache.h | 5
- b/include/asterisk/format_cap.h | 15
- b/include/asterisk/frame.h | 40
- b/include/asterisk/http_websocket.h | 14
- b/include/asterisk/iostream.h | 14
- b/include/asterisk/logger.h | 426
- b/include/asterisk/manager.h | 2
- b/include/asterisk/max_forwards.h | 1
- b/include/asterisk/message.h | 31
- b/include/asterisk/mixmonitor.h | 5
- b/include/asterisk/netsock2.h | 42
- b/include/asterisk/options.h | 7
- b/include/asterisk/parking.h | 5
- b/include/asterisk/res_audiosocket.h | 87
- b/include/asterisk/res_fax.h | 3
- b/include/asterisk/res_pjsip.h | 106
- b/include/asterisk/res_pjsip_presence_xml.h | 5
- b/include/asterisk/res_pjsip_pubsub.h | 23
- b/include/asterisk/res_pjsip_session.h | 27
- b/include/asterisk/res_pjsip_session_caps.h | 82
- b/include/asterisk/res_stir_shaken.h | 117
- b/include/asterisk/rtp_engine.h | 111
- b/include/asterisk/say.h | 4
- b/include/asterisk/serializer.h | 85
- b/include/asterisk/slin.h | 5
- b/include/asterisk/sorcery.h | 27
- b/include/asterisk/stasis.h | 3
- b/include/asterisk/stasis_app.h | 10
- b/include/asterisk/stasis_channels.h | 8
- b/include/asterisk/stream.h | 463
- b/include/asterisk/strings.h | 53
- b/include/asterisk/taskprocessor.h | 9
- b/include/asterisk/utf8.h | 188
- b/include/asterisk/utils.h | 59
- b/include/asterisk/vector.h | 50
- b/main/Makefile | 1
- b/main/abstract_jb.c | 178
- b/main/acl.c | 105
- b/main/app.c | 26
- b/main/ast_expr2.c | 1
- b/main/ast_expr2.y | 1
- b/main/asterisk.c | 41
- b/main/asterisk.exports.in | 1
- 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 | 20
- b/main/backtrace.c | 9
- b/main/bridge.c | 15
- b/main/bridge_channel.c | 62
- b/main/cdr.c | 15
- b/main/channel.c | 204
- b/main/channel_internal_api.c | 12
- b/main/cli.c | 261
- b/main/codec_builtin.c | 8
- b/main/config.c | 28
- b/main/conversions.c | 51
- b/main/core_local.c | 112
- b/main/core_unreal.c | 141
- b/main/data_buffer.c | 2
- b/main/db.c | 3
- b/main/dial.c | 14
- b/main/dns_core.c | 75
- b/main/dns_srv.c | 6
- b/main/dns_txt.c | 127
- b/main/enum.c | 104
- b/main/event.c | 17
- b/main/features.c | 38
- b/main/file.c | 46
- b/main/format_cache.c | 8
- b/main/format_cap.c | 24
- b/main/frame.c | 160
- b/main/http.c | 62
- b/main/indications.c | 6
- b/main/iostream.c | 14
- b/main/logger.c | 150
- b/main/manager.c | 210
- b/main/media_cache.c | 47
- b/main/message.c | 55
- b/main/named_acl.c | 9
- b/main/options.c | 11
- b/main/pbx.c | 35
- b/main/pbx_variables.c | 31
- b/main/rtp_engine.c | 137
- b/main/say.c | 968 -
- b/main/serializer.c | 189
- b/main/sorcery.c | 46
- b/main/stasis.c | 53
- b/main/stasis_cache.c | 10
- b/main/stasis_channels.c | 7
- b/main/stasis_state.c | 298
- b/main/strcompat.c | 94
- b/main/stream.c | 432
- b/main/taskprocessor.c | 219
- b/main/tcptls.c | 33
- b/main/translate.c | 8
- b/main/utf8.c | 380
- b/main/utils.c | 50
- b/makeopts.in | 6
- b/menuselect/configure | 14
- b/menuselect/menuselect.c | 14
- b/menuselect/menuselect.h | 2
- b/pbx/pbx_dundi.c | 21
- b/res/Makefile | 2
- b/res/ari/ari_model_validators.c | 59
- b/res/ari/ari_model_validators.h | 23
- b/res/ari/config.c | 10
- b/res/ari/resource_bridges.c | 1
- b/res/ari/resource_bridges.h | 2
- b/res/ari/resource_channels.c | 245
- b/res/ari/resource_channels.h | 46
- b/res/ari/resource_events.c | 10
- b/res/parking/parking_bridge.c | 36
- b/res/parking/parking_bridge_features.c | 14
- b/res/parking/res_parking.h | 5
- b/res/res_agi.c | 20
- b/res/res_ari_bridges.c | 7
- b/res/res_ari_channels.c | 147
- b/res/res_ari_events.c | 2
- b/res/res_audiosocket.c | 345
- b/res/res_audiosocket.exports.in | 4
- 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 | 7
- b/res/res_config_pgsql.c | 2
- b/res/res_corosync.c | 564
- b/res/res_fax.c | 37
- b/res/res_http_websocket.c | 97
- b/res/res_musiconhold.c | 472
- b/res/res_phoneprov.c | 6
- b/res/res_pjsip.c | 492
- b/res/res_pjsip/config_system.c | 8
- b/res/res_pjsip/config_transport.c | 17
- b/res/res_pjsip/location.c | 25
- b/res/res_pjsip/pjsip_configuration.c | 209
- b/res/res_pjsip/pjsip_message_filter.c | 40
- b/res/res_pjsip/pjsip_options.c | 7
- b/res/res_pjsip/pjsip_resolver.c | 30
- b/res/res_pjsip_acl.c | 20
- b/res/res_pjsip_caller_id.c | 3
- b/res/res_pjsip_config_wizard.c | 7
- b/res/res_pjsip_dialog_info_body_generator.c | 80
- b/res/res_pjsip_diversion.c | 7
- b/res/res_pjsip_dlg_options.c | 3
- b/res/res_pjsip_dtmf_info.c | 1
- b/res/res_pjsip_empty_info.c | 1
- b/res/res_pjsip_endpoint_identifier_ip.c | 102
- b/res/res_pjsip_history.c | 4
- b/res/res_pjsip_logger.c | 451
- b/res/res_pjsip_messaging.c | 58
- b/res/res_pjsip_mwi.c | 339
- b/res/res_pjsip_nat.c | 110
- b/res/res_pjsip_notify.c | 22
- b/res/res_pjsip_one_touch_record_info.c | 1
- b/res/res_pjsip_outbound_registration.c | 66
- b/res/res_pjsip_path.c | 1
- b/res/res_pjsip_pubsub.c | 148
- b/res/res_pjsip_refer.c | 8
- b/res/res_pjsip_registrar.c | 80
- b/res/res_pjsip_rfc3326.c | 1
- b/res/res_pjsip_sdp_rtp.c | 283
- b/res/res_pjsip_session.c | 667 -
- b/res/res_pjsip_session.exports.in | 1
- b/res/res_pjsip_session/pjsip_session_caps.c | 164
- b/res/res_pjsip_stir_shaken.c | 330
- b/res/res_pjsip_t38.c | 40
- b/res/res_pjsip_transport_websocket.c | 4
- b/res/res_realtime.c | 56
- b/res/res_resolver_unbound.c | 6
- b/res/res_rtp_asterisk.c | 1100 +
- b/res/res_sorcery_config.c | 14
- b/res/res_sorcery_memory_cache.c | 18
- b/res/res_speech.c | 7
- b/res/res_srtp.c | 8
- b/res/res_stasis.c | 57
- b/res/res_stasis_playback.c | 4
- b/res/res_statsd.c | 35
- b/res/res_stir_shaken.c | 1663 ++
- b/res/res_stir_shaken.exports.in | 6
- b/res/res_stir_shaken/certificate.c | 388
- b/res/res_stir_shaken/certificate.h | 119
- b/res/res_stir_shaken/curl.c | 199
- b/res/res_stir_shaken/curl.h | 73
- b/res/res_stir_shaken/general.c | 286
- b/res/res_stir_shaken/general.h | 111
- b/res/res_stir_shaken/stir_shaken.c | 119
- b/res/res_stir_shaken/stir_shaken.h | 55
- b/res/res_stir_shaken/store.c | 202
- b/res/res_stir_shaken/store.h | 37
- b/res/stasis/app.c | 15
- b/res/stasis/command.c | 2
- b/res/stasis/control.c | 14
- b/res/stasis/messaging.c | 11
- b/res/stasis/stasis_bridge.c | 16
- b/res/stasis/stasis_bridge.h | 3
- b/rest-api-templates/make_ari_stubs.py | 2
- b/rest-api-templates/res_ari_resource.c.mustache | 2
- b/rest-api/api-docs/bridges.json | 9
- b/rest-api/api-docs/channels.json | 143
- b/rest-api/api-docs/endpoints.json | 20
- b/rest-api/resources.json | 2
- b/tests/CI/buildAsterisk.sh | 2
- b/tests/CI/gates.jenkinsfile | 12
- b/tests/CI/periodics-daily.jenkinsfile | 11
- b/tests/CI/publishAsteriskDocs.sh | 4
- b/tests/CI/ref_debug.jenkinsfile | 9
- b/tests/CI/unittests.jenkinsfile | 9
- b/tests/CI/universal-asterisk-nongerrit.jenkinsfile | 452
- b/tests/test_conversions.c | 153
- b/tests/test_data_buffer.c | 2
- b/tests/test_locale.c | 12
- b/tests/test_res_pjsip_session_caps.c | 176
- b/tests/test_res_rtp.c | 516
- b/tests/test_scope_trace.c | 126
- b/tests/test_stasis.c | 2
- b/tests/test_stream.c | 2
- b/tests/test_strings.c | 39
- b/tests/test_taskprocessor.c | 78
- b/tests/test_utils.c | 11
- b/third-party/pjproject/Makefile | 3
- b/third-party/pjproject/configure.m4 | 199
- b/third-party/pjproject/patches/0011-sip_inv_patch.patch | 39
- b/third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch | 39
- b/third-party/pjproject/patches/0050-fix-race-parallel-build.patch | 72
- b/third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch | 28
- b/third-party/pjproject/patches/config_site.h | 4
- b/third-party/pjproject/pjproject-2.10.tar.bz2.md5 | 2
- b/third-party/versions.mak | 2
- b/utils/Makefile | 6
- b/utils/astman.c | 2
- b/utils/db1-ast/hash/ndbm.c | 3
- b/utils/db1-ast/include/db.h | 1
- b/utils/extconf.c | 10
- b/utils/muted.c | 11
- third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
- third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch | 53
- third-party/pjproject/patches/0030-ssl-regression-fix.patch | 105
- third-party/pjproject/patches/0031-transport-regression-fix.patch | 187
- third-party/pjproject/pjproject-2.9.tar.bz2.md5 | 2
- 392 files changed, 29541 insertions(+), 9926 deletions(-)
projects / thirdparty / asterisk.git / commitdiff
