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-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-certified/16.8-cert1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-certified/16.8-cert1</h3><h3 align="center">Date: 2020-04-30</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 release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-certified/16.3-cert1.</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%">53 Sean Bright <sean.bright@gmail.com><br/>51 George Joseph <gjoseph@digium.com><br/>25 Kevin Harwell <kharwell@digium.com><br/>25 Joshua Colp <jcolp@sangoma.com><br/>18 Asterisk Development Team <asteriskteam@digium.com><br/>14 Alexei Gradinari <alex2grad@gmail.com><br/>13 Joshua C. Colp <jcolp@sangoma.com><br/>10 Ben Ford <bford@digium.com><br/>6 Corey Farrell <git@cfware.com><br/>6 Richard Mudgett <rmudgett@digium.com><br/>5 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>5 Jaco Kroon <jaco@uls.co.za><br/>5 Tzafrir Cohen <tzafrir.cohen@xorcom.com><br/>4 sungtae kim <pchero21@gmail.com><br/>4 Abhay Gupta <abhay@avissol.com><br/>4 Chris-Savinovich <csavinovich@digium.com><br/>3 Salah Ahmed <txrubel@gmail.com><br/>3 Pascal Cadotte Michaud <pcm@wazo.io><br/>3 Dan Cropp <dan@amtelco.com><br/>3 Igor Goncharovsky <igor.goncharovsky@gmail.com><br/>3 Guido Falsi <madpilot@FreeBSD.org><br/>2 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>2 Pirmin Walthert <infos@nappsoft.ch><br/>2 Walter Doekes <walter+asterisk@wjd.nu><br/>2 Torrey Searle <torrey@voxbone.com><br/>2 lvl <digium@lvlconsultancy.nl><br/>1 Matthew Fredrickson <creslin@digium.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 Thomas Arimont (license 5525)<br/>1 Nasir Iqbal <nasir@ictinnovations.com><br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Stas Kobzar <stas@modulis.ca><br/>1 Francesco Castellano <francesco.castellano@messagenet.it><br/>1 Jonathan Rose <jrose@digium.com><br/>1 Antoni Goldstein <action@gdevel.com><br/>1 Morten Tryfoss <morten@tryfoss.no><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Michael Goryainov<br/>1 Jean Aunis <jean.aunis@prescom.fr><br/>1 Leonid Fainshtein <leonid.fainshtein@xorcom.com><br/>1 Lucas Mendes <lucas.mendes@wearespindle.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Holger Hans Peter Freyther <holger@moiji-mobile.com><br/>1 cmaj <chris@penguinpbx.com><br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Kirsty Tyerman <kirsty.tyerman@boeing.com><br/>1 snuffy <snuffy22@gmail.com><br/>1 Asterisk Team <root@DIGIUM1.digium.internal><br/>1 Alexander Anikin <may213@yandex.ru><br/></td><td width="33%">1 tests/test_utils.c.<br/>1 Abhay Gupta<br/></td><td width="33%">16 Joshua C. Colp <jcolp@digium.com><br/>9 Ross Beer <ross.beer@voicehost.co.uk><br/>8 Kevin Harwell <kharwell@digium.com><br/>6 Ross Beer<br/>5 Frederic LE FOLL <frederic.lefoll@c-s.fr><br/>5 George Joseph <gjoseph@digium.com><br/>4 Pascal Cadotte Michaud <pascal.cadotte@gmail.com><br/>4 Abhay Gupta <abhay@avissol.com><br/>4 Salah Ahmed <txrubel@gmail.com><br/>4 cmaj <chris@penguinpbx.com><br/>4 sungtae kim <pchero21@gmail.com><br/>3 Dan Cropp <dan@amtelco.com><br/>3 Sean Bright <sean.bright@gmail.com><br/>3 Guido Falsi <madpilot@freebsd.org><br/>3 Dan Cropp<br/>3 nappsoft <infos@nappsoft.ch><br/>2 Walter Doekes <walter+asterisk@wjd.nu><br/>2 Joshua Elson <joshelson@gmail.com><br/>2 Bernhard Schmidt<br/>2 Corey Farrell <git@cfware.com><br/>2 Stas Kobzar <stas@modulis.ca><br/>2 Bernhard Schmidt <berni@birkenwald.de><br/>2 Ruddy G <plugworld@micnes.com><br/>2 Gregory Massel <greg@csurf.co.za><br/>2 Alexei Gradinari <alex2grad@gmail.com><br/>2 Jonathan Harris <lardconcepts@gmail.com><br/>2 Torrey Searle <tsearle@gmail.com><br/>1 Oleksandr Natalenko<br/>1 Martin Tomec <tomec.martin@gmail.com><br/>1 AvayaXAsterisk<br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Steven Wheeler <swheeler@usinternet.com><br/>1 Byron Clark <bclark@getjive.com><br/>1 candrews <candrews@integralblue.com><br/>1 Jean Aunis - Prescom <jean.aunis@prescom.fr><br/>1 Yoooooo Ha <n1906374c@e.ntu.edu.sg><br/>1 kevin@phoneburner.com<br/>1 Gil Richard <grichard@intertalksystems.com><br/>1 Oleksandr Natalenko <oleksandr@natalenko.name><br/>1 Marian Piater <marian.piater@voipsun.cz><br/>1 Michael Goryainov <gms4nlt@gmail.com><br/>1 Niksa Baldun <niksa.baldun@gmail.com><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Mark <mark@wrapped.cx><br/>1 Steven Wheeler<br/>1 Dirk Wendland<br/>1 Bryan Nelson <bnelson@fluentstream.com><br/>1 Sam Banks <sam.banks.nz@gmail.com><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Speed Dial Dave <speed_dial_dave@gmx.com><br/>1 Richard Kenner <kenner@gnat.com><br/>1 Sébastien Duthil <sduthil@wazo.community><br/>1 Joshua C. Colp<br/>1 Sébastien Duthil<br/>1 Aheliotech <phones@aheliotech.com><br/>1 Jim Van Meggelen<br/>1 Robert Sutton<br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Kevin Flyn<br/>1 Janu<br/>1 Frank Matano <ftalarico99@gmail.com><br/>1 pasandev <pasandev@ymail.com><br/>1 Cédric Bassaget<br/>1 Kevin Flyn <kevflynn69@gmail.com><br/>1 Dan Jenkins <dan@nimbleape.com><br/>1 Luke-Jr <luke-jr+digiumbugs@utopios.org><br/>1 Robert Sutton <rsutton@noojee.com.au><br/>1 Jeremiah Gadd <jeremygadd@gmail.com><br/>1 Michael <ringo@vianet.ca><br/>1 Boris P. Korzun <drtr0jan@yandex.ru><br/>1 Kilburn <kilburna@gmail.com><br/>1 Alexander Traud<br/>1 Joeran Vinzens<br/>1 Dennis <dennis.buteyn@xorcom.com><br/>1 test011 <tanus@tanus.org><br/>1 Joeran Vinzens <vinzens@sipgate.de><br/>1 Jim Van Meggelen <jim.vanmeggelen@clearlycore.com><br/>1 Kirill Katsnelson<br/>1 Kirsty Tyerman <kirsty.tyerman@boeing.com><br/>1 Lucas Mendes <lucas.mendes@wearespindle.com><br/>1 Timothy Vanderaerden <timothy.vanderaerden@optimise-group.be><br/>1 Janu <mdp.87.cat@gmail.com><br/>1 Florian Floimair <f.floimair@commend.com><br/>1 Michael Maier <m1278468@mailbox.org><br/>1 Daniel <depeee@gmail.com><br/>1 Dan Jenkins<br/>1 Robin Leffmann <robin@stolendata.net><br/>1 Mitch Claborn<br/>1 Antoni Goldstein <action@gdevel.com><br/>1 David Lee<br/>1 Dmitry Shubin <dssaster@comita.ru><br/>1 klaus3000 <ramon@pernau.at><br/>1 Maciej Michno <maciej.michno@xtb.com><br/>1 Dmitry Shubin<br/>1 Gil Richard<br/>1 Kevin Reeves <kevin@phoneburner.com><br/>1 Niklas Larsson <niklas@tese.se><br/>1 Dmitry Svyatogorov <ds@vo-ix.ru><br/>1 Jean-Denis Girard<br/>1 Christoph Moench-Tegeder <cmt@burggraben.net><br/>1 Maciej Michno<br/>1 the CC variable, instead of unconditionally<br/>1 Stas Kobzar<br/>1 Francesco Castellano <francesco.castellano@messagenet.it><br/>1 Cedric BASSAGET <cedric@oceanet.com><br/>1 Ted G <tgwaste@gmail.com><br/>1 Frank Matano<br/>1 David M. Lee <dlee@digium.com><br/>1 vijay kumar <vijaykumar@drishti-soft.com><br/>1 Niklas Larsson<br/>1 Andrey V. T. <avt1203@gmail.com><br/>1 Francois Blackburn <fblackburn@wazo.io><br/>1 Juan Martin <jmartin79@yandex.com><br/>1 Richard Kenner<br/>1 Abhay Gupta<br/>1 Ian Jones <tech@iljones.net><br/>1 Jean-Denis Girard <jd.girard@sysnux.pf><br/>1 lvl <digium@lvlconsultancy.nl><br/>1 Ted G<br/>1 Marin Odrljin <marin@maxcom.hr><br/>1 Morten Tryfoss <morten@tryfoss.no><br/>1 Andrew Siplas <andrew@asiplas.net><br/>1 Vyrva Igor <vigor1710@yandex.ru><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 Luke-Jr <luke-jr+digiumbugs@utopios.org><br/>1 abelbeck <lonnie@abelbeck.com><br/>1 Jonathan Harris<br/>1 Nasir Iqbal <nasir@ictinnovations.com><br/>1 Chris Savinovich <csavinovich@digium.com><br/>1 Kirill Katsnelson <kkm@pobox.com><br/>1 Eliel Sardañons<br/>1 Sean Bright<br/>1 Kirsty Tyerman<br/>1 Cyril Ramière <cyril.ramiere@ino.global><br/>1 Jørgen H <asterisk.org@hovland.cx><br/>1 Niksa Baldun<br/>1 dennis <dennis@arena1.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=8cdaa93e658a46e7baf6b606468b5e2c88a0133b">[8cdaa93e65]</a> Ben Ford -- chan_sip.c: Prevent address change on unauthenticated SIP request.</li>
-</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28465">ASTERISK-28465</a>: Broken SDP can cause a segfault in a T.38 reINVITE<br/>Reported by: Francesco Castellano<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c59df17a55a4f91a05df3a833652e04f0853cf6">[6c59df17a5]</a> Francesco Castellano -- chan_sip: Handle invalid SDP answer to T.38 re-invite</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=7574be5110e049a44b8c8ead52cd1c2a5d442afa">[7574be5110]</a> George Joseph -- manager.c: Prevent the Originate action from running the Originate app</li>
-</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28447">ASTERISK-28447</a>: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash<br/>Reported by: Gil Richard<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2126dc30219b110e5f2870c7cca61676d6848933">[2126dc3021]</a> George Joseph -- res_pjsip_messaging: Check for body in in-dialog message</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=965df3c228d49bcde3503e0482f3c831dcbf6c77">[965df3c228]</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=8894a5645228c002500c7657f434ffbdd2a77700">[8894a56452]</a> lvl -- app_senddtmf: Add receive mode to AMI Action PlayDTMF</li>
-</ul><br><h4>Category: Core/Jitterbuffer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28533">ASTERISK-28533</a>: func_jitterbuffer: Add support for video synchronization<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6647be69acc1d662efe4c458a7522e5a70bc8276">[6647be69ac]</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=f69da94fab0cab6968292c0b2f3b1eb46fddf510">[f69da94fab]</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=37dcdd485a96e15ddcb5a0b49ffdca422a2a0592">[37dcdd485a]</a> Martin Tomec -- func_curl.c: Support custom http headers</li>
-</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28320">ASTERISK-28320</a>: Added ARI resource /ari/channels/{channelid}/rtp_statistics<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbc13b1f1f4b0bd007f71c947dee0cf5afa86d01">[bbc13b1f1f]</a> sungtae kim -- res/res_ari: Added ARI resource /ari/channels/{channelId}/rtp_statistics</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=b478f46d59114733f716c41356585af54c10c1f8">[b478f46d59]</a> sungtae kim -- res_musiconhold: Added unregister realtime moh class</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28375">ASTERISK-28375</a>: res_pjsip: New configuration setting to allow disabling norefersub<br/>Reported by: Dan Cropp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eca8c440d2df8a314029d486e5496f736b8ac4aa">[eca8c440d2]</a> Dan Cropp -- res_pjsip: Added a norefersub configuration setting</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=f8b0c2c9331e55820dec08bf3eb2aef5b8e8947c">[f8b0c2c933]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add port matching support</li>
-</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28375">ASTERISK-28375</a>: res_pjsip: New configuration setting to allow disabling norefersub<br/>Reported by: Dan Cropp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eca8c440d2df8a314029d486e5496f736b8ac4aa">[eca8c440d2]</a> Dan Cropp -- res_pjsip: Added a norefersub configuration setting</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=fb984eda40f9753a5f98b59044386664129bbb52">[fb984eda40]</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=7db5f5df6ae6b807849277e169624a70e45a2077">[7db5f5df6a]</a> Kevin Harwell -- res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions</li>
-</ul><br><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28348">ASTERISK-28348</a>: Failed to initialize OOH323 endpoint-OOH323 Disabled<br/>Reported by: Dmitry Shubin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eec16b8e99de3da54000e03121bf889d854c0755">[eec16b8e99]</a> Alexander Anikin -- chan_ooh323: fix h323 log file path</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=cf5b7f3a0b1003ed43ac1554b5077afe07ed3bd1">[cf5b7f3a0b]</a> Michael Cargile -- app_amd: Fixed timeout issue</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28419">ASTERISK-28419</a>: app_amd: Does not work with silence suppression<br/>Reported by: Nasir Iqbal<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52a3d4a761e7ca438a96f58f606b8efb442dce8a">[52a3d4a761]</a> Nasir Iqbal -- app_amd: issue with silence suppression fixed</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28143">ASTERISK-28143</a>: app_amd: Infinite loop on silent calls <br/>Reported by: Abhay Gupta<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d214a36236131cceefa2e4f0438f8b3c3614bb0">[1d214a3623]</a> Abhay Gupta -- app_amd: Fix infinite loop on silent calls</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=aa06c6ea297ccb82ae30a11f66f523dc1b18652c">[aa06c6ea29]</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=c8cf3ad389ffbcf96e3079fc776a27da737f79b7">[c8cf3ad389]</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-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=3d22ecb1e2b02f812d5a6854e769b12727341d5e">[3d22ecb1e2]</a> Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient space for worst case NACK.</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=7167fd6d463ad046dacd890dd4fe6a3ccc34e026">[7167fd6d46]</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=be4c6f3f35485a90cb5c1c402fcfa5fd4c4c94af">[be4c6f3f35]</a> George Joseph -- cdr_mysql: Fix missing use of 'my_bool' with MySql >= 8.0.1</li>
-</ul><br><h4>Category: Applications/app_queue</h4><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=5fded77e7f7c1a62a02d9e99054fc5e95f385d0e">[5fded77e7f]</a> Sean Bright -- app_queue: Deprecate the QueueMemberPause.Reason field</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=161e762742e147c16b040618923a4797ce8b286d">[161e762742]</a> Walter Doekes -- app_queue: Fix old confusing comment about when the members are called</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=0c07a7ee00374665fcc82bd170383fc165f92f45">[0c07a7ee00]</a> Corey Farrell -- app_record: Do not hang up if beep audio is missing</li>
-</ul><br><h4>Category: Applications/app_transfer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26968">ASTERISK-26968</a>: chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer<br/>Reported by: Dan Cropp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4896703b95b8d34301f4a919b3e488a7f042bb2">[f4896703b9]</a> Dan Cropp -- chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS</li>
-</ul><br><h4>Category: Applications/app_voicemail</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=293600724dd285b305974520754a2be0707cf163">[293600724d]</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=e379fe48e1ffebd8eb3a4fc250a2f5978a922c78">[e379fe48e1]</a> Sean Bright -- app_voicemail: Set globals to default values when voicemail.conf missing</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27935">ASTERISK-27935</a>: app_voicemail: emailbody per user can't contain commas<br/>Reported by: Sébastien Duthil<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d0a8334e4f5119cdf06a08d73aa3012609936f1b">[d0a8334e4f]</a> Sean Bright -- app_voicemail: Don't split mailbox options on comma</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=ff180a5bfc3090485ee7b2746ce7bee58e754746">[ff180a5bfc]</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=293600724dd285b305974520754a2be0707cf163">[293600724d]</a> Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail</li>
-</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28637">ASTERISK-28637</a>: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.<br/>Reported by: Frederic LE FOLL<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31173f6586fcfaf304b18b827bf95713eb1ee564">[31173f6586]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
-</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28618">ASTERISK-28618</a>: bridge_softmix: hold not cleared when joining a softmix bridge<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3891a953cf4f242f6de9d28bf410e3a6672fee1c">[3891a953cf]</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=1b452ebb51b6d29e9d48d877d2aa54f67dfc5a18">[1b452ebb51]</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=aa06c6ea297ccb82ae30a11f66f523dc1b18652c">[aa06c6ea29]</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=495bc77a6a409ef2975a14b270126cc764767039">[495bc77a6a]</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=6c54bd704e920a51ff0b537165a6a131c6316be3">[6c54bd704e]</a> Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28435">ASTERISK-28435</a>: cdr_pgsql: Unix socket doesn't work<br/>Reported by: Dmitry Svyatogorov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2621aa190a167f49e9bee9324db9d0540ee4b3a">[c2621aa190]</a> Chris-Savinovich -- cdr_pgsql: fix error in connection string</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=9895e94dba8cd4b9985f5c73562b94be17e823e1">[9895e94dba]</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=faf353e93109a4ebf5c15989dbd81e2a20ab3287">[faf353e931]</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=448b8c9bc278ed79f1f4ec9b139050556c2e636a">[448b8c9bc2]</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=c6b17b521231dde9da890d95ee705c93953bab8c">[c6b17b5212]</a> Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28457">ASTERISK-28457</a>: [patch] Fix crash in chan_dahdi on 32-bit systems caused by ASTERISK-28317<br/>Reported by: abelbeck<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da1db4f842b097f9c2e6183a5659cda49b2be3ea">[da1db4f842]</a> Chris-Savinovich -- chan_dahdi.c: crash in chan_dahdi</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28427">ASTERISK-28427</a>: new mwi.h include missing from some dahdi source files, causes build failure<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86fb72c4d08961f9db01818a55f0fa499d465d36">[86fb72c4d0]</a> Guido Falsi -- chan_dahdi: add missing include.</li>
-</ul><br><h4>Category: Channels/chan_local</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28399">ASTERISK-28399</a>: channel.c: Exceptionally long queue length queuing<br/>Reported by: Abhay Gupta<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a0fa514433eed4d8681abb7f0fae92071dba435">[9a0fa51443]</a> Abhay Gupta -- stasis: Hangup channel for Local channel No such extension error</li>
-</ul><br><h4>Category: Channels/chan_pjsip</h4><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=992dcdf780bf30f9e0a40b32a48f2e0dff8f5996">[992dcdf780]</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=63b8664bfa7cb971a27940fdb09aa8b98ad71d9b">[63b8664bfa]</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=c73aaa276033df3872374b4a35a965cff96600e0">[c73aaa2760]</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=a66848c92f5ccc4df6aac90ef340534b30290b69">[a66848c92f]</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=19565e0b905d08b29c6b3c32837f672ab61bc4ac">[19565e0b90]</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=49e1346185c6b3a3d99de447f4c0e8fa91a6de2c">[49e1346185]</a> Joshua Colp -- chan_pjsip: Relock correct channel during "fax" redirect.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26968">ASTERISK-26968</a>: chan_pjsip: Transfer() does not result in TRANSFERSTATUS reflecting SIP response to transfer<br/>Reported by: Dan Cropp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4896703b95b8d34301f4a919b3e488a7f042bb2">[f4896703b9]</a> Dan Cropp -- chan_pjsip: Transmit REFER waits for the REFER result setting TRANSFERSTATUS</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28444">ASTERISK-28444</a>: chan_pjsip: Peer IP for SSL handshake errors not logged<br/>Reported by: Bernhard Schmidt<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2db5173b88aca5138ec6949d1806c13ecaada922">[2db5173b88]</a> George Joseph -- pjproject_bundled: Add peer information to most SSL/TLS errors</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25371">ASTERISK-25371</a>: Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts to generate hangup event<br/>Reported by: Abhay Gupta<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72f26aa8eb9ec6b07ac790e83df41a0ffdf1af0e">[72f26aa8eb]</a> Abhay Gupta -- chan_pjsip.c: Check for channel and session to not be NULL in hangup</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27994">ASTERISK-27994</a>: PJSIP: Early media ringback not indicated after Progress()<br/>Reported by: Gregory Massel<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de82bdd746f20df959caf04cc6d5cb160cfadb35">[de82bdd746]</a> Alexei Gradinari -- pjsip: replace 180 by 183 if SDP negotiation has completed</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28379">ASTERISK-28379</a>: pjsip: show channelstats incorrect information output<br/>Reported by: Vyrva Igor<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ece29db9bdf604a01f7ab3a50a1a7195cd2baee0">[ece29db9bd]</a> Joshua Colp -- res_rtp_asterisk: Fix sequence number cycling and packet loss count.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28371">ASTERISK-28371</a>: chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info<br/>Reported by: Salah Ahmed<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7e5b4b8616a5fd287f1ede281864c09631787958">[7e5b4b8616]</a> Salah Ahmed -- chan_pjsip: DTMF Mode auto_info fallback lead to both inband and info</li>
-</ul><br><h4>Category: Channels/chan_sip/General</h4><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=82a870c8c7e4f328b2bc4c19fed1515fc5f2b60c">[82a870c8c7]</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-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=055737d645d56c61c246b6342f2ae4eddd2420b2">[055737d645]</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-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=31173f6586fcfaf304b18b827bf95713eb1ee564">[31173f6586]</a> Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28282">ASTERISK-28282</a>: AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in chan_sip)<br/>Reported by: Walter Doekes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=64d25d36fb4c1cdba3e8b4791462c4805715aee8">[64d25d36fb]</a> Walter Doekes -- sched: Don't allow ast_sched_del to deadlock ast_sched_runq from same thread</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28362">ASTERISK-28362</a>: strtok_r() makes gcc compile warning<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8961d9ca8b84225cae3e1632d8612ab2eda72637">[8961d9ca8b]</a> Ben Ford -- build: Fix compiler warnings/errors.</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=31027f33db11dbdd5c49b06f7f6811d8367862c9">[31027f33db]</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-26006">ASTERISK-26006</a>: Show offending IP for TLS setup failures in logs<br/>Reported by: Oleksandr Natalenko<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dc61e41fa93ae9da0299ba24aa2b27f187e9a18">[0dc61e41fa]</a> George Joseph -- tcptls.c: Add peer hostname and port to some error messages</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=1b452ebb51b6d29e9d48d877d2aa54f67dfc5a18">[1b452ebb51]</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-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=92261d60c876cb0be63eb75390525bd1732a5db8">[92261d60c8]</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=bf527810efba18baabf84fcd931a2cb9fc65d707">[bf527810ef]</a> Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cdbb9800e311e4d9ca98215ba14348fe5d9e224e">[cdbb9800e3]</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=efecc9d139e9107fe45e2ee7e96a615fc7fb3dcd">[efecc9d139]</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=a78758d0a219c291d065203d56da6b48d350ceab">[a78758d0a2]</a> Sean Bright -- config.c: Skip UTF-8 BOMs if present when reading config files</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=b9b50774f54a495d3a814c1acc9c0e3e7cccc9c4">[b9b50774f5]</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=b8e635916fb5650a44d69e7851192207cc5a08ae">[b8e635916f]</a> Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid typo</li>
-</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28487">ASTERISK-28487</a>: compile menuselect on gentoo<br/>Reported by: Kilburn<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8399211eaf55fa27b845ff089fbeac5f3cf535ef">[8399211eaf]</a> Sean Bright -- menuselect: Fix curses build on Gentoo Linux</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28392">ASTERISK-28392</a>: The no-partial-inlining flag isn't passed to the bundled pjproject or jansson builds<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=543d487746403b3ed8059175c666ac65d435df32">[543d487746]</a> George Joseph -- build: Pass --fno-partial-inlining to third-party when appropriate</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28374">ASTERISK-28374</a>: latest asterisk unconditionally launch gcc --version, even if the compiler is different<br/>Reported by: Guido Falsi<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4dcfa8d1277b23a933152e37db342ff3e6c07c51">[4dcfa8d127]</a> Guido Falsi -- core/buildsystem: check the actual compiler being version</li>
-</ul><br><h4>Category: Core/Channels</h4><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=3297df5a61397243745eb2665105fbab763db7ae">[3297df5a61]</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=2691ee7e106ee537e6347c4e69a3ef7e417667e6">[2691ee7e10]</a> Joshua Colp -- AST-2019-005 - translate: Don't assume all frames will have a src.</li>
-</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23756">ASTERISK-23756</a>: setvar directive when used in template and a child of said template, results in duplicate variable names<br/>Reported by: Michael Goryainov<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=518b6bfb5c370a015d5ddd0b67d53bb6da67a5f7">[518b6bfb5c]</a> Michael Goryainov -- channels: Allow updating variable value</li>
-</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28498">ASTERISK-28498</a>: cel / cdr: Event times may be incorrect<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6350f4e278569c084f519eff3302a5f5eebec355">[6350f4e278]</a> Joshua Colp -- cdr / cel: Use event time at event creation instead of processing.</li>
-</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26006">ASTERISK-26006</a>: Show offending IP for TLS setup failures in logs<br/>Reported by: Oleksandr Natalenko<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0dc61e41fa93ae9da0299ba24aa2b27f187e9a18">[0dc61e41fa]</a> George Joseph -- tcptls.c: Add peer hostname and port to some error messages</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=6bb14150c4b19407ad2d166ee6f0bc94415e9d21">[6bb14150c4]</a> Kevin Harwell -- various modules: json integer overflow</li>
-</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28553">ASTERISK-28553</a>: stasis.c: Crash during unload<br/>Reported by: Kevin Harwell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ce1e0714ba9127aad89b7042ea206efac430f9f7">[ce1e0714ba]</a> Joshua Colp -- stasis: Pass bumped topic_all reference to proxy_dtor.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28353">ASTERISK-28353</a>: stasis: Crash at shutdown when statistics enabled<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d35a30a3f3de6760653636bdd1bc8b794c9b358">[8d35a30a3f]</a> Ben Ford -- stasis: Fix crash at shutdown.</li>
-</ul><br><h4>Category: Core/Streams</h4><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=e013f502b1b5abda167206a3ec2dbe50ff59a585">[e013f502b1]</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=83390327b2036b4e4d34c4e94fcbed0b057a8b03">[83390327b2]</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-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=04c81f9748d7d380d0fe2f957b34afc492185acf">[04c81f9748]</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=8d87fef5a1701eff8929cdb4c0686e6339aab253">[8d87fef5a1]</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=4cf32f2578cc65360ae951bd607a842694037876">[4cf32f2578]</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=2d2b28bfa4f725c6b4568b62f63329a1662bb11f">[2d2b28bfa4]</a> Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=174e6426aa8a8c26d83c5e06cfbc77277c2f5934">[174e6426aa]</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=e39ddb1cb19ccfa609ada6b04ef23d0a4b825e6b">[e39ddb1cb1]</a> Sean Bright -- func_env: Prevent FILE() from reading garbage at end-of-file</li>
-</ul><br><h4>Category: Functions/func_odbc</h4><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=e54299cd3e2b93937d26ef866afc2f65095507f9">[e54299cd3e]</a> Boris P. Korzun -- func_odbc: acf_odbc_read() and cli_odbc_read() unicode support</li>
-</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28609">ASTERISK-28609</a>: Memory Leak in res_rtp_asterisk.c<br/>Reported by: Ted G<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8af0dea0c785a40b18559bb4eb1d26aa46faa1a5">[8af0dea0c7]</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=e627db79282034143551cfae309dde16418a3348">[e627db7928]</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=bd96a0b79d18c6904cb7d3342073002ccc820cb0">[bd96a0b79d]</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=d4766a82a28cce3c068fcb45cbb34fba0328b28c">[d4766a82a2]</a> Kevin Harwell -- srtp: Fix possible race condition, and add NULL checks</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28412">ASTERISK-28412</a>: GCC 9 catches more string formatting issues<br/>Reported by: George Joseph<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7734476c6080af0e84b5e1b49a6c0020686f360">[e7734476c6]</a> George Joseph -- Fixes for GCC 9</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28319">ASTERISK-28319</a>: musl: Crash on startup when loading modules<br/>Reported by: Sebastian Kemper<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8ec4de7501d9ea340a950107c56513cced9bc97e">[8ec4de7501]</a> Sebastian Kemper -- loader: support for permanent dlopen()</li>
-</ul><br><h4>Category: PBX/General</h4><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=f5a1e8b04d2349c1df3c05e4f30c2e845b4cbd70">[f5a1e8b04d]</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=64692a3c7204e2644209d6a9fbf45ca825ed1aaa">[64692a3c72]</a> George Joseph -- sig_pri: Fix deadlock caused by sig_pri_queue_hangup</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20182">ASTERISK-20182</a>: Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIfTime application causes unexpected behavior<br/>Reported by: Janu<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f91262272eda3021555be9bfabeac88520698548">[f91262272e]</a> Sean Bright -- pbx.c: Properly parse labels with leading digits</li>
-</ul><br><h4>Category: PBX/pbx_ael</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17799">ASTERISK-17799</a>: AEL reload causes loss of control in a macro<br/>Reported by: Kirill Katsnelson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=116dc9c9b3d3ad4db8ccd510ff6fb598a6b16768">[116dc9c9b3]</a> Sean Bright -- res_ael: Create consistent label names across reloads</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18593">ASTERISK-18593</a>: AEL for loops use Macro app and pipe delimiter<br/>Reported by: Luke-Jr<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea3109beaabdc96e6bb64d962334fc1ca2eec75f">[ea3109beaa]</a> Sean Bright -- res_ael: Use Gosub in for loop expressions</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-14939">ASTERISK-14939</a>: AEL parsers does not find existing label<br/>Reported by: klaus3000<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71c7864d1d7e7aafda12cb4cb4ed8a2f02d8f4a1">[71c7864d1d]</a> Sean Bright -- res_ael: Fix pattern matching against literal '+'</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=051455900506060fbad6fced5beda281afc6e79d">[0514559005]</a> Sean Bright -- pbx: Prevent Realtime switch crash on invalid priority</li>
-</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-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=1627e8eddc378b7df54f298ca0f4a89294a10864">[1627e8eddc]</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=6e22e1213eb0ed1baff0734dcfcc8f23e355da16">[6e22e1213e]</a> Joshua Colp -- res_ari_events: Add module reference when a WebSocket is open.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26718">ASTERISK-26718</a>: ARI: Bridge destroying doesn't work as expected<br/>Reported by: Marin Odrljin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f599ebd29ee01d10383d93222032f216f1b66ae6">[f599ebd29e]</a> Holger Hans Peter Freyther -- stasis: Call callbacks when imparting fails</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=ccaf735d1f6879defa3633ba20621108e8619c32">[ccaf735d1f]</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=ccaf735d1f6879defa3633ba20621108e8619c32">[ccaf735d1f]</a> Sean Bright -- res_calendar: Resolve memory leak on calendar destruction</li>
-</ul><br><h4>Category: Resources/res_config_sqlite3</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28477">ASTERISK-28477</a>: Crash when not specifying "dbfile" in res_config_sqlite3.conf<br/>Reported by: Dennis<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28654308efcc677f61534c6e9d883db3fb396eca">[28654308ef]</a> Sean Bright -- res_config_sqlite3: Only join threads that we started</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28478">ASTERISK-28478</a>: Crash performing "core reload" with modified res_config_sqlite3.conf<br/>Reported by: Dennis<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28654308efcc677f61534c6e9d883db3fb396eca">[28654308ef]</a> Sean Bright -- res_config_sqlite3: Only join threads that we started</li>
-</ul><br><h4>Category: Resources/res_fax</h4><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=d17bbcb9f179e6bd4df78a744d44d6ee836f3f5f">[d17bbcb9f1]</a> Kevin Harwell -- res_fax: wrap v21 detected Asterisk initiated negotiation with config option</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27981">ASTERISK-27981</a>: res_fax: Fax session leak with fax gatewaying<br/>Reported by: pasandev<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0a574253e9ac2f91cf2860ae2a18adf32784e56">[e0a574253e]</a> Alexei Gradinari -- res_fax: fix segfault on inactive "reserved" fax session</li>
-</ul><br><h4>Category: Resources/res_http_websocket</h4><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=47ba42f4a054d97f0c71027594b0e2868317a19f">[47ba42f4a0]</a> Sean Bright -- websocket: Consider pending SSL data when waiting for socket input</li>
-</ul><br><h4>Category: Resources/res_indications</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28391">ASTERISK-28391</a>: res_indications: Crash requesting autocomplete on indications cli command<br/>Reported by: Lucas Mendes<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=daed593cfa33b07e818a38193feb51e3caf709b5">[daed593cfa]</a> Lucas Mendes -- res_indications: Fix indications remove command autocomplete</li>
-</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28631">ASTERISK-28631</a>: res_parking: Doesn't park when parkee and parker are the same<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c91b28c72d7641619681e79afbd9cd9213caaab2">[c91b28c72d]</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=b9bbf394497a5d45d69ee5890c9f68dff23c590f">[b9bbf39449]</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=e7320bbbf002595e3244daa3dad3868bfdd30992">[e7320bbbf0]</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-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=3d22ecb1e2b02f812d5a6854e769b12727341d5e">[3d22ecb1e2]</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-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=68ce999351105fea32bb4ff22e01d2ec4603045f">[68ce999351]</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=1f10ca76da93e9f939fe2be7393dd0fbe2c73f08">[1f10ca76da]</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=789c51ac8afd7616705395816669f67e4238694d">[789c51ac8a]</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=c2b135729caad2c08526fabeca80f7e519aca2a2">[c2b135729c]</a> Joshua Colp -- res_pjsip: Fix multiple of the same contact in "pjsip show contacts".</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=8d87fef5a1701eff8929cdb4c0686e6339aab253">[8d87fef5a1]</a> Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames</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=17e71b6abe705d3078e2b905680f9b334904c122">[17e71b6abe]</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=3f12cd77113b04799e9a036edddbfb8a420e3052">[3f12cd7711]</a> Kevin Harwell -- res_pjsip_mwi: use an ao2_global object for mwi containers</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=0a56edca4d4d0a446bafc90314c897844f3f5737">[0a56edca4d]</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=bb19e7feb5b33c8696a3a57be900f75fe36c009f">[bb19e7feb5]</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=3cd57aaff2eff5f84f2a5b8e0223ee78361fd178">[3cd57aaff2]</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=789c51ac8afd7616705395816669f67e4238694d">[789c51ac8a]</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=41cd1ff454294148a5d072b59d36d86af1e1954f">[41cd1ff454]</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=939e18d63eee6c228446c4b14021ba8940761574">[939e18d63e]</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=8318b05f2513adf97aa6ff9ebcafafc5737ebc53">[8318b05f25]</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-28402">ASTERISK-28402</a>: res_pjsip_registrar: SEGV in registrar_find_contact<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5002169d6aa4c0949535c79dec7264e5694f80d1">[5002169d6a]</a> George Joseph -- res_pjsip: Check return from pjsip_parse_uri calls</li>
-</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28659">ASTERISK-28659</a>: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them<br/>Reported by: nappsoft<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=186c4e9b36e66bb9526acb2a845a52dc18c3b844">[186c4e9b36]</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-28458">ASTERISK-28458</a>: res_pjsip_sdp_rtp: Remove unused variable<br/>Reported by: Michael Maier<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=83aba363fe57fd5192205586828bada78384dd9b">[83aba363fe]</a> Kevin Harwell -- res_pjsip_sdp_rtp: Remove unused variable</li>
-</ul><br><h4>Category: Resources/res_pjsip_session</h4><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=aa04c3f49b0bc0cab745c213f690a5718421ea22">[aa04c3f49b]</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=d515dea9c6bedd9ad09ed546c9bd2e0d822b4256">[d515dea9c6]</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=186c4e9b36e66bb9526acb2a845a52dc18c3b844">[186c4e9b36]</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=fbc80db350ee006cdebcb113a6daf60f95c8851d">[fbc80db350]</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=19565e0b905d08b29c6b3c32837f672ab61bc4ac">[19565e0b90]</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=efef44985d8db83d23cd66d09e56780266066668">[efef44985d]</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=fbe18165d5155c3c687b40fbd13c8d7175ee3bfe">[fbe18165d5]</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-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=22bc8a71680e4099ffd3ccff7b3fe33d26291c36">[22bc8a7168]</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=fef8a04aadf759bf08f12827de18f970ae7e3e8c">[fef8a04aad]</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-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=4458b7a52ad65b8fac22015d3b9c44f799710a96">[4458b7a52a]</a> Joshua C. Colp -- res_rtp_asterisk: Improve video performance in certain networks.</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=43ca35b8312c9c9be740bda4b961262e751c5a54">[43ca35b831]</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=eb54e381fde42afcb038dae2ae41da727d088be0">[eb54e381fd]</a> Joshua Colp -- res_rtp_asterisk: Always return provided DTLS packet length.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28018">ASTERISK-28018</a>: IP Fragmentation happening instead of DTLS fragmentation on handshake server hello certificate<br/>Reported by: vijay kumar<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82789aafd6bd428b5112aa3d028d8ce4098acc1e">[82789aafd6]</a> Joshua Colp -- res_rtp_asterisk: Add support for DTLS packet fragmentation.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28421">ASTERISK-28421</a>: Wrong type used for timestamp in res_rtp_asterisk<br/>Reported by: Morten Tryfoss<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9351aa3f0e2c45db7c2379a94688531b6e674765">[9351aa3f0e]</a> Morten Tryfoss -- res_rtp_asterisk: timestamp should be unsigned instead of signed int</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28255">ASTERISK-28255</a>: res_rtp_asterisk: REMB RTCP packet sending may be incorrect<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94eeba61473aa2cdbb43eefb94e60e8fe18e02ca">[94eeba6147]</a> Kevin Harwell -- bridge_softmix: use a float type to store the internal REMB bitrate</li>
-</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28423">ASTERISK-28423</a>: ARI causes STASIS Deadlock<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42c51263b9ec8a0bb349af39e28ff51ed3d682ee">[42c51263b9]</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=40c49ec64f04183ef808467f63297d8acc8df54e">[40c49ec64f]</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=dd82ebecd34bd18fe6569b950402e23786a05664">[dd82ebecd3]</a> George Joseph -- stasis.c: Use correct topic name in stasis_topic_pool_delete_topic</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-27756">ASTERISK-27756</a>: bridge: Failure to impart a channel results in bad data causing crash<br/>Reported by: Abhay Gupta<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39c5188becf105eacf58d19d876b9358b7a9f434">[39c5188bec]</a> Abhay Gupta -- stasis: Only place stasis created and dialed channels into dial bridge.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26718">ASTERISK-26718</a>: ARI: Bridge destroying doesn't work as expected<br/>Reported by: Marin Odrljin<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f599ebd29ee01d10383d93222032f216f1b66ae6">[f599ebd29e]</a> Holger Hans Peter Freyther -- stasis: Call callbacks when imparting fails</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=04c81f9748d7d380d0fe2f957b34afc492185acf">[04c81f9748]</a> Sean Bright -- res_statsd: Document that res_statsd does nothing on its own</li>
-</ul><br><h4>Category: Tests/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17695">ASTERISK-17695</a>: 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with '-' in them<br/>Reported by: test011<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9b7a64cbf0e0eed0b5c83be25ad7febfa39f965a">[9b7a64cbf0]</a> Sean Bright -- pbx.c: Ignore dashes in extensions when using extenpatternmatchnew</li>
-</ul><br><h4>Category: Utilities/aelparse</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18593">ASTERISK-18593</a>: AEL for loops use Macro app and pipe delimiter<br/>Reported by: Luke-Jr<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea3109beaabdc96e6bb64d962334fc1ca2eec75f">[ea3109beaa]</a> Sean Bright -- res_ael: Use Gosub in for loop expressions</li>
-</ul><br><h4>Category: Utilities/conf2ael</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18593">ASTERISK-18593</a>: AEL for loops use Macro app and pipe delimiter<br/>Reported by: Luke-Jr<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea3109beaabdc96e6bb64d962334fc1ca2eec75f">[ea3109beaa]</a> Sean Bright -- res_ael: Use Gosub in for loop expressions</li>
-</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28574">ASTERISK-28574</a>: pjproject fails to build on 16.6.0, works on 16.5<br/>Reported by: Niklas Larsson<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb6e0d36aedfb0b58e167bdc3fc4ec27c9f3ddb7">[cb6e0d36ae]</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=c8cc530726f5f81c99bc4181c8b8c39a1842913f">[c8cc530726]</a> Dan Cropp -- pjproject: Configurable setting for cnonce to include hyphens or not</li>
-</ul><br><h3>Improvement</h3><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=5622df0a94a9d7762d07fa08d88c40fc2a566d28">[5622df0a94]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28401">ASTERISK-28401</a>: app_confbridge: Add *_all remb behavior variants<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d861ebdca83f14cc7e74804bf3e99ba2d0cc806a">[d861ebdca8]</a> Joshua Colp -- app_confbridge: Add "all" variants of REMB behavior.</li>
-</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28363">ASTERISK-28363</a>: Millisecond-resolution call stats including PDD in channel variables<br/>Reported by: Antoni Goldstein<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d6b37e2926121dd707f0f700431e6a788e7adba5">[d6b37e2926]</a> Antoni Goldstein -- app_dial.c: RINGTIME, PROGRESSTIME and ms resolution dial timings</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=68855f6a7bd39fb9362a735a5a25ea3964c4d388">[68855f6a7b]</a> Sean Bright -- Revert "app_voicemail: Cleanup stale lock files on module load"</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20207">ASTERISK-20207</a>: Asterisk should clear out any .lock files in the voice mail directory on startup.<br/>Reported by: Steven Wheeler<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34b9b65098783b867c24367191da2a3bcb93ee62">[34b9b65098]</a> Sean Bright -- 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=b41a8fd0f30888fae7b5fdf2bd6bd7cacf16a302">[b41a8fd0f3]</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=5fccf6d1c6a05ef048bc96bb8fc5d28a657b956c">[5fccf6d1c6]</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=5fccf6d1c6a05ef048bc96bb8fc5d28a657b956c">[5fccf6d1c6]</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=5fccf6d1c6a05ef048bc96bb8fc5d28a657b956c">[5fccf6d1c6]</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=5622df0a94a9d7762d07fa08d88c40fc2a566d28">[5622df0a94]</a> Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28401">ASTERISK-28401</a>: app_confbridge: Add *_all remb behavior variants<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d861ebdca83f14cc7e74804bf3e99ba2d0cc806a">[d861ebdca8]</a> Joshua Colp -- app_confbridge: Add "all" variants of REMB behavior.</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=a7692ce2f4cf1d2a591cb94e176fda16199113c6">[a7692ce2f4]</a> Richard Mudgett -- app_chanisavail.c: Simplify dialplan using ChanIsAvail.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=144b774b850c20202bcc2784530258e0800eb4ed">[144b774b85]</a> Richard Mudgett -- app_dial.c: Simplify dialplan using Dial.</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2780be334d44a0124f82e8c6e05660c1d32aa2fe">[2780be334d]</a> Richard Mudgett -- app_page.c: Simplify dialplan using Page.</li>
-</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28111">ASTERISK-28111</a>: build: CHANGES/UPGRADE are irritating to work with.<br/>Reported by: Corey Farrell<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ef404fef91316b49f49dd396383b2cff4fad9beb">[ef404fef91]</a> Ben Ford -- build: Revise CHANGES and UPGRADE.txt handling.</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=f85631cf824e236b8c97ab9d48d56bfaeae109e9">[f85631cf82]</a> Florian Floimair -- core: Add H.265/HEVC passthrough support</li>
-</ul><br><h4>Category: Core/HTTP</h4><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=a2a4e1026cf2de03d9978c320b37652aca3b45b6">[a2a4e1026c]</a> Sean Bright -- http: Add ability to disable /httpstatus URI</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=5fccf6d1c6a05ef048bc96bb8fc5d28a657b956c">[5fccf6d1c6]</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-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=60fd1322d795f51f027019dc77ce4e56324bc3ec">[60fd1322d7]</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=2666a5e11155cfd11ee7f1bbd538a65f65179242">[2666a5e111]</a> Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling correetions.</li>
-</ul><br><h4>Category: PBX/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28264">ASTERISK-28264</a>: Added topic_all container<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5815597a21fa1e9efea503ff3a86f8bc82c967ca">[5815597a21]</a> sungtae kim -- stasis.c: Added topic_all container</li>
-</ul><br><h4>Category: PBX/pbx_dundi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28234">ASTERISK-28234</a>: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi<br/>Reported by: Kirsty Tyerman<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1c84709b868583d5d030b1ba093c1228edc0779">[a1c84709b8]</a> Kirsty Tyerman -- pbx_dundi: added IPv4/IPv6 dual bind support to DUNDi</li>
-</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28343">ASTERISK-28343</a>: Added app_name, app_data to channel type<br/>Reported by: sungtae kim<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5a318f148a8d7d2e4c3d336f7ea314eecaec728">[d5a318f148]</a> sungtae kim -- main/json.c: Added app_name, app_data to channel type</li>
-</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28542">ASTERISK-28542</a>: [patch] add the ability for asterisk to generate on-hold re-invites<br/>Reported by: Torrey Searle<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a933c3adc43ac49525c718bbafa209fea362b57">[9a933c3adc]</a> Torrey Searle -- channel/chan_pjsip: add dialplan function for music on hold</li>
-</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28602">ASTERISK-28602</a>: res_pjsip_outbound_registration: Maximum retries reached<br/>Reported by: Daniel<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=95bc698b85a34d907ece21379df3669a1e627a76">[95bc698b85]</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-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=5fccf6d1c6a05ef048bc96bb8fc5d28a657b956c">[5fccf6d1c6]</a> Joshua C. Colp -- bridging: Add better support for adding/removing streams.</li>
-</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28400">ASTERISK-28400</a>: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5023f02b2dea8bb09cad4962ed57cef8504a5720">[5023f02b2d]</a> Joshua Colp -- rtp: Add support for transport-cc in receiver direction.</li>
-</ul><br><h4>Category: Resources/res_pjsip_session</h4><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=bdf4d159fdb609efb739ee25e87e4e4bd40faefb">[bdf4d159fd]</a> Joshua C. Colp -- res_pjsip_session: Apply intention behind requested formats.</li>
-</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28400">ASTERISK-28400</a>: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for transport-cc<br/>Reported by: Joshua C. Colp<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5023f02b2dea8bb09cad4962ed57cef8504a5720">[5023f02b2d]</a> Joshua Colp -- rtp: Add support for transport-cc in receiver direction.</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: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28161">ASTERISK-28161</a>: Removal of Previous Patch Causes PJSIP Timer Issues<br/>Reported by: Ross Beer<ul>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=33ed2e1bb86bf19852ec8872ef0ff7273e264103">[33ed2e1bb8]</a> Joshua Colp -- pjproject-bundled: Add upstream timer fixes</li>
-<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4cd2a9706e6dde796aeb3c912afe5449b99e6ba">[d4cd2a9706]</a> Sean Bright -- pjproject: Add timer patch from pjproject r5934</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=fe3dc091b57bf16ba62e185fe05f77069062a3b1">fe3dc091b5</a></td><td>Joshua Colp</td><td>Revert "res_rtp_asterisk: Free payload when error on insertion to data buffer"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a75317ce247b913d1cb92b9ffa47a49aa56b172a">a75317ce24</a></td><td>Joshua Colp</td><td>Revert "res_rtp_asterisk: Resolve loop when receive buffer is flushed"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=81d34554abc050305ffbbf046358deb70f64bd56">81d34554ab</a></td><td>Asterisk Development Team</td><td>Update for certified/16.8-cert1-rc5</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25e2274e48e263a5359f0947296f4124065535ea">25e2274e48</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=2af88e7ca2c94ab5b5c4ea30e422bbcfc4919f5e">2af88e7ca2</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=3d317239d5e94f07d387b31c46a6733cbc43e5ef">3d317239d5</a></td><td>Asterisk Development Team</td><td>Update for certified/16.8-cert1-rc4</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=096db196620afe4df2857be61b59ad3d48e00ff1">096db19662</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for certified/16.8-cert1-rc4</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c9cd68126152bae26d42f5b9ce8811ddf1eda4d8">c9cd681261</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=904f6b7c2f0ad658dffb28cb55db4bff5e9310a2">904f6b7c2f</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=7ecbcdf22f1aa02151a945698a1dce641ace777e">7ecbcdf22f</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=d1bb76a27d2b8b4e4d32e77e8090997400f1d46d">d1bb76a27d</a></td><td>Asterisk Development Team</td><td>Update for certified/16.8-cert1-rc3</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b922e387d186b979773c01e394f22154c639587c">b922e387d1</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for certified/16.8-cert1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b8157645f8c5f8599f160cd3374d2763564b55f">7b8157645f</a></td><td>Asterisk Development Team</td><td>Update for certified/16.8-cert1-rc2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=476bbcf3a3a8439c469ab31677cc87bbfd2fb214">476bbcf3a3</a></td><td>Asterisk Development Team</td><td>Update for certified/16.8-cert1-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3a0387fc01dd4019d1462d220c83dbf8ac00d74">b3a0387fc0</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for certified/16.8-cert1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7416703f04f12eb583a3427a3f64d06951c18c6e">7416703f04</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=443230f5d5cb2eff29f0b5a2a2d1cae6e58ff800">443230f5d5</a></td><td>George Joseph</td><td>Asterisk Certified 16.8 Preparation</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40e331ff905249bd3a35a106c40254618c6ed69f">40e331ff90</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=8b0922a101ae5edb8e761981829d9fc74096037f">8b0922a101</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=d436f568583184a13aa46349af5a3f0907087b44">d436f56858</a></td><td>Asterisk Development Team</td><td>Update for 16.8.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=126beb3e6c6b647a14cf9bbfb1d18659684b6d0a">126beb3e6c</a></td><td>Joshua Colp</td><td>REVERT: 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=bfe9e1b2e7a489b7eb49a98d290f2e3a68a34dca">bfe9e1b2e7</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=c92e2bb09f2b12a0bba06c7d95c194f097a0bba3">c92e2bb09f</a></td><td>Asterisk Development Team</td><td>Update for 16.8.0-rc2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b7b813eb3498223daf9a2ac883aadafec6fa65cb">b7b813eb34</a></td><td>Asterisk Development Team</td><td>Update for 16.8.0-rc1</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb1ec0498d10f8824ccd85146606808fb1b54dc7">eb1ec0498d</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.8.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a7aaca9eaa698309f0ee837cd15590f343d0417d">a7aaca9eaa</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=f49517efb9ce9c7d39d71effb1d1cd01ffc7b3d6">f49517efb9</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=13fa33588fbe106c3d79f372f62637f13f014194">13fa33588f</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=b92b0469ff17273fbf97a7c33ec2573fafc4d09e">b92b0469ff</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=de078debab0b7aef82511a81f7af740eb0a3af76">de078debab</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=11753d94d8799f18a2e13b660e6b596197284797">11753d94d8</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=00e745066c4a7e0ad0cfa624dc6098cc16aff570">00e745066c</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=198f4cbdbf034814fbba03b1921ffe710263e828">198f4cbdbf</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=efa13eb0a076cc9958681d6f07288d059305472a">efa13eb0a0</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=9240fcd8bbf06642da9218abdef35148f9cb2784">9240fcd8bb</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.7.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77941efad9063cf1cb0d8e3382d47ced2d05ac0c">77941efad9</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=9c29c3fb3e85811a75c841b6ed93c928dbf75409">9c29c3fb3e</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=5c20cc4c3aa0684ff2a81513c85165718ca09d87">5c20cc4c3a</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=43d4c0e3c9ac27ea0b3cd49085e72465b63e3014">43d4c0e3c9</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=80199cd67f48642dc5e2a6b37a2452ee74b757ca">80199cd67f</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=bd3cb1b300adcac6562557608de4d703320ecea9">bd3cb1b300</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=9eb86a8110a220e2e7e05754a87fe3388aee681a">9eb86a8110</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.6.2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d075d8913b69fa8445edf90d3153c964666eb654">d075d8913b</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=ce8a23fdf966dc6824678f3cb722753db06baa7a">ce8a23fdf9</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=919bc0c7be94bd323f76bbba286aa63540f21424">919bc0c7be</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=61a3e7e79b68b89fdf48aa9371c55190bb6878d2">61a3e7e79b</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=30c0af7257eb11a69efbd4edfe1a45ec27759e1e">30c0af7257</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=6f0a69c51a25fa63405b991d7c20baa229eef96a">6f0a69c51a</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=6be18dfb721196c903d0c4a76ca008d93ba80037">6be18dfb72</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=bfd0e05e5996fc751e9342a7e6f319a4704a8a6c">bfd0e05e59</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=ef2eb064b2f8f9453b937d2a3c6683421b6ee2a2">ef2eb064b2</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=ed12715cbe1f9cb7a68a41efc2503c23132a2352">ed12715cbe</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=867c8b2879a0de7875a456f677ada6774798386a">867c8b2879</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=afc10c25ac911a6fe75c98354fe41f2c4c166209">afc10c25ac</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=931ef77e2102955453297b310157e8f9b02955d6">931ef77e21</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=04f7d136d8ae4454cf403dc9bb24cf0fbbab4509">04f7d136d8</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=d146ec7e835291185dfbd52c286a006ebbf0d114">d146ec7e83</a></td><td>Asterisk Team</td><td>Revert "Update CHANGES and UPGRADE.txt for 16.6.0-rc2"</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85c7326d0841e7cf09e121153ddf0d5ae71b3b1d">85c7326d08</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=827dd754b2140baceef3fde7e3126f011de5f661">827dd754b2</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=a0844a9b9bbe313c524bca82550e7ec1ed2fe928">a0844a9b9b</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.6.0-rc2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ff11c2f0054192caf18e058090442924266268a">9ff11c2f00</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=cd51f5b876669c018b2e5a99ac58c999514456ec">cd51f5b876</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=5ea667e03acfb6ab03010b756ad45e9b678a67d0">5ea667e03a</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=fec6e1bd87cb32682b78a62f745ec936df2f2ae5">fec6e1bd87</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=9c5a8066a6599366468c6b87f3badb21d90fac01">9c5a8066a6</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=5b8c1ed0d38c8872440e4b84a67a9e21d9b0c938">5b8c1ed0d3</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=76d4a42ae1b99c472ca62f9916eef2dbaa13e708">76d4a42ae1</a></td><td>Corey Farrell</td><td>res_pjsip_mwi: Remove inappropriate topic unreference.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=256db889f2e4de01846b20a23cf39b6f03b94753">256db889f2</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=9f304170f6004c2fc4dcc23bef16194a816260c8">9f304170f6</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=a95cef71409a2c791f6184e14a3fb8608067b9fe">a95cef7140</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=688908fe7a6d26cfd3ef3b7213986103b749dd4e">688908fe7a</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.6.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=922d3e02df4450b4911b45212dbadc655db2e205">922d3e02df</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=d566314e38d47313af73dc5150c259323e930528">d566314e38</a></td><td>George Joseph</td><td>ARI: External Media</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a321225fa443d1fb823ad992d72c2ef825df67ef">a321225fa4</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=78d00c277cebad260577024a59227ef7fc0920e7">78d00c277c</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=821b7561f85b5fcbc337b4dfcec85709812c56e3">821b7561f8</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=aaaa1695ca76b659a9131ceaa86ae3fa4fb3aca0">aaaa1695ca</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=c00a010fe8e4cac63f09406a8c7ff97166f5f773">c00a010fe8</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=6407ccd2d9887754d51b38de6d635e42fd775611">6407ccd2d9</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=f82d0b74fd870b94609c196c11bdc9c4d5b9663b">f82d0b74fd</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=51fd43206ba2655233f660f92da2424ffa2ab874">51fd43206b</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=92066b87469080831c678d16305f4043b1e26861">92066b8746</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=db9684ad1e2a74329da744aede8bd8c5f293f41c">db9684ad1e</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=2641081caa7f1c9755a782509d5a46a1c59e6b3a">2641081caa</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=97183769028a8791f31d441e3495bb444bcc22a1">9718376902</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=0ebfc4a19ddd6d2d53840ca18d7cd53a5a572b81">0ebfc4a19d</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=d6af1acb8c21eb1552663e5c335aea93373e7fed">d6af1acb8c</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=05cf9c99128cff8a83711624f2e78c049957b54e">05cf9c9912</a></td><td>George Joseph</td><td>loader.c: Fix possible SEGV when a module fails to register</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06780d2bc4c8304de2709e5ceb32f011559878e4">06780d2bc4</a></td><td>George Joseph</td><td>CI: Don't enable non-core modules in Certified branches</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3814faf8486214d57d9ffad9cf8e8a39254772c5">3814faf848</a></td><td>Leonid Fainshtein</td><td>openr2(6/6): Set hangup cause</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06515707dff10a8016723a81bc6132ced28f21cb">06515707df</a></td><td>Tzafrir Cohen</td><td>openr2(5/6): added cli command -- mfcr2 destroy link <index></td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=93a093f6c4e01357ed598f426bb0233e9f9f9c4d">93a093f6c4</a></td><td>Tzafrir Cohen</td><td>openr2(4/6): added new cli command -- mfcr2 show links</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a45cfefb77e7f58254d48f65f61b57d457e66102">a45cfefb77</a></td><td>Tzafrir Cohen</td><td>openr2(3/6): Convert r2links to standard Asterisk AST_LIST*</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec6e88592a94ac0e16a8c313df0584e052fe3dce">ec6e88592a</a></td><td>Tzafrir Cohen</td><td>openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out)</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b6df814a6dc731acc5f3acc8f4e730425256af3">7b6df814a6</a></td><td>Tzafrir Cohen</td><td>openr2(1/6): bugfix in configuration saving</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=356f4256cc53d5fa6d9ced1b92f0f2f80e1ae166">356f4256cc</a></td><td>George Joseph</td><td>CI: Add cleanWs to cleanup steps in jenkinsfiles</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d694692032b85e7360c625e8240dbae4951d53c">9d69469203</a></td><td>Rodrigo Ramírez Norambuena</td><td>README.md: Update year</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c86c0973ff61f8379a729076c6bbb747384d41cf">c86c0973ff</a></td><td>George Joseph</td><td>CI: Add install-headers to the install make targets</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb831a18d15a9c1434df757a23d34bd3f488af99">cb831a18d1</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.5.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f62d9013c10cf5c677a61926696502639abc2fe0">f62d9013c1</a></td><td>George Joseph</td><td>Build: Add separate header install/uninstall targets</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88ea395c333acea93791a21b948f361408e2f5a9">88ea395c33</a></td><td>Kevin Harwell</td><td>manager: Log AMI actions</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17560292371766a4dc0ba236f84c94a1e5d6212c">1756029237</a></td><td>Joshua Colp</td><td>res_rtp_asterisk: Move where DTLS MTU variable is defined.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31d755e8050f0dbe2b8a214625674f04767c1f87">31d755e805</a></td><td>George Joseph</td><td>sig_pri: Address gcc9 issues</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=01712bbdc950f4d1efce4251c360e7db793809e4">01712bbdc9</a></td><td>George Joseph</td><td>CI: New way to determnine libdir</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ee2f01f62572aeec728e5413a89527e24ba8cfb">1ee2f01f62</a></td><td>George Joseph</td><td>chan_dahdi: Address gcc9 issues</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b77318a2cc4746529380271cd9da4c62b819b90">8b77318a2c</a></td><td>Alexei Gradinari</td><td>translate.c do not log WARNING on empty audio frame</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ccc92b6ecb6f40df20cceff2d1e2703bf7230c04">ccc92b6ecb</a></td><td>George Joseph</td><td>app_confbridge: Attended transfer event fixup</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=694097ee68c25e8cb58f44e68e09e19283870010">694097ee68</a></td><td>Sean Bright</td><td>pjproject: Update to 2.9 release</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86cd77ec0a22b380a73b9d859779181ae6e13bff">86cd77ec0a</a></td><td>Alexei Gradinari</td><td>app_attended_transfer: new application AttendedTransfer</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6321b559b9ef0881690233907320294a9beea3ce">6321b559b9</a></td><td>Alexei Gradinari</td><td>res_fax: gateway sends T.38 request to both endpoints if V.21 detected</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2c07acecaa34434fd989c972a2432b084d1789d">d2c07aceca</a></td><td>Asterisk Development Team</td><td>Update CHANGES and UPGRADE.txt for 16.4.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e77704f45c829c605f2eb28ce72ef49892cfe393">e77704f45c</a></td><td>Alexei Gradinari</td><td>res_fax: add channel name to CLI 'fax show session'</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec74fd56a7c22c858c4b7f8876b5610aab09dd3e">ec74fd56a7</a></td><td>Ben Ford</td><td>build: Fix file format in CHANGES-staging.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=db5bc0fabfb68c87e04770ba2a5ea57be7bfb117">db5bc0fabf</a></td><td>Alexei Gradinari</td><td>app_blind_transfer: new application BlindTransfer</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9516fb64c9d59437c6527a23f8d5168e0d7eaf0e">9516fb64c9</a></td><td>Alexei Gradinari</td><td>app_readexten: new option 'p' to stop reading on '#' key</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79b15d0b302b776172125af30e41a929d8a282fc">79b15d0b30</a></td><td>George Joseph</td><td>res_rtp_asterisk: Add ability to propose local address in ICE</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=941dead08d1a9a7f1a5932109b663d54edfc5467">941dead08d</a></td><td>Ben Ford</td><td>pjsip_options.c: Allow immediate qualifies for new contacts.</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=edc3e0df1aa75f935369e337238fb340352472b5">edc3e0df1a</a></td><td>Kevin Harwell</td><td>conversions.c: Add conversions for largest max sized integer</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e3a758975da6c7c8a7b8bc0dcf6c2d33b5080b28">e3a758975d</a></td><td>Kevin Harwell</td><td>mwi core: Move core MWI functionality into its own files</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e281911667449368237ffdee0db096e0126ca55c">e281911667</a></td><td>George Joseph</td><td>ARI: Bump non-breaking version number to 4.0.2</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7487fc88d27e0cece072c48ef3790d44b33a4564">7487fc88d2</a></td><td>George Joseph</td><td>res_remb_modifier: Propertly initialize bitrate to 0.0</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=022e784b7a2004defbe149e0a871d60045b48885">022e784b7a</a></td><td>Sean Bright</td><td>res_mwi_devstate: Specify AST_MODFLAG_LOAD_ORDER to enable load priority</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=18fe583d128a86e9cf6a9c47545fc5e24f988636">18fe583d12</a></td><td>George Joseph</td><td>CI: Move test group config files to Jenkins</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=898765d9192f556f9f583115d095cd72a03f5393">898765d919</a></td><td>George Joseph</td><td>ARI: Run 'make ari-stubs'</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=acfbfef8adf16cbe6912d1fe34d92c15ec57b7b1">acfbfef8ad</a></td><td>Alexei Gradinari</td><td>res_pjsip: Fix transport_states ref leak</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85bbb7a3e852593fcb00be91c1c01f0cccb150ac">85bbb7a3e8</a></td><td>Chris-Savinovich</td><td>config.c: Fix a crash in extconfig parsing</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1464a6b80fd39a7a2a2757ff0b8195431bd66163">1464a6b80f</a></td><td>George Joseph</td><td>CI: Add --no-dev-mode option to buildAsterisk.sh</td></tr>
-<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae1aeb930e9117fa8c73f40b7557c9e5aebf38a6">ae1aeb930e</a></td><td>Matthew Fredrickson</td><td>res/res_rtp_asterisk: Enable rxjitter calculation for video</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>UPGRADE-1.2.txt | 218
-UPGRADE-1.4.txt | 497
-UPGRADE-1.6.txt | 277
-UPGRADE-1.8.txt | 343
-UPGRADE-10.txt | 92
-UPGRADE-11.txt | 280
-UPGRADE-12.txt | 478
-UPGRADE-13.txt | 399
-UPGRADE-14.txt | 115
-UPGRADE-15.txt | 63
-asterisk-certified-16.3-cert1-summary.html | 7965 ----
-asterisk-certified-16.3-cert1-summary.txt |19102 ----------
-b/.gitreview | 2
-b/.version | 2
-b/CHANGES | 270
-b/ChangeLog | 4334 ++
-b/Makefile | 30
-b/Makefile.rules | 9
-b/README-SERIOUSLY.bestpractices.md | 4
-b/README.md | 2
-b/UPGRADE.txt | 2611 +
-b/addons/cdr_mysql.c | 18
-b/addons/chan_ooh323.c | 19
-b/addons/chan_ooh323.h | 1
-b/addons/ooh323c/src/ooh323ep.c | 6
-b/addons/ooh323c/src/ooh323ep.h | 2
-b/apps/app_adsiprog.c | 2
-b/apps/app_agent_pool.c | 8
-b/apps/app_amd.c | 32
-b/apps/app_attended_transfer.c | 144
-b/apps/app_blind_transfer.c | 138
-b/apps/app_cdr.c | 8
-b/apps/app_chanisavail.c | 137
-b/apps/app_chanspy.c | 3
-b/apps/app_confbridge.c | 163
-b/apps/app_dial.c | 138
-b/apps/app_dictate.c | 4
-b/apps/app_followme.c | 12
-b/apps/app_minivm.c | 4
-b/apps/app_mixmonitor.c | 13
-b/apps/app_page.c | 30
-b/apps/app_playback.c | 9
-b/apps/app_queue.c | 20
-b/apps/app_readexten.c | 14
-b/apps/app_record.c | 3
-b/apps/app_senddtmf.c | 13
-b/apps/app_softhangup.c | 2
-b/apps/app_voicemail.c | 580
-b/apps/confbridge/conf_config_parser.c | 35
-b/apps/confbridge/confbridge_manager.c | 27
-b/apps/confbridge/include/confbridge.h | 14
-b/asterisk-certified-16.8-cert1-rc5-summary.html | 16
-b/asterisk-certified-16.8-cert1-rc5-summary.txt | 101
-b/bridges/bridge_native_rtp.c | 173
-b/bridges/bridge_simple.c | 198
-b/bridges/bridge_softmix.c | 377
-b/bridges/bridge_softmix/include/bridge_softmix_internal.h | 2
-b/cdr/cdr_pgsql.c | 26
-b/cel/cel_pgsql.c | 2
-b/channels/chan_console.c | 4
-b/channels/chan_dahdi.c | 480
-b/channels/chan_dahdi.h | 19
-b/channels/chan_iax2.c | 29
-b/channels/chan_mgcp.c | 1
-b/channels/chan_motif.c | 9
-b/channels/chan_pjsip.c | 200
-b/channels/chan_rtp.c | 19
-b/channels/chan_sip.c | 43
-b/channels/chan_skinny.c | 1
-b/channels/chan_unistim.c | 175
-b/channels/pjsip/cli_commands.c | 13
-b/channels/pjsip/dialplan_functions.c | 65
-b/channels/pjsip/include/dialplan_functions.h | 25
-b/channels/sig_analog.c | 7
-b/channels/sig_pri.c | 33
-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 | 5
-b/configs/samples/confbridge.conf.sample | 6
-b/configs/samples/dundi.conf.sample | 6
-b/configs/samples/extconfig.conf.sample | 1
-b/configs/samples/func_odbc.conf.sample | 8
-b/configs/samples/http.conf.sample | 10
-b/configs/samples/musiconhold.conf.sample | 23
-b/configs/samples/pjsip.conf.sample | 6
-b/configs/samples/rtp.conf.sample | 4
-b/configure | 192
-b/configure.ac | 36
-b/contrib/ast-db-manage/README.md | 1
-b/contrib/ast-db-manage/config/versions/3a094a18e75b_pjsip_add_norefersub.py | 39
-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 | 21
-b/contrib/realtime/postgresql/postgresql_config.sql | 27
-b/contrib/scripts/ast_coredumper | 417
-b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
-b/contrib/valgrind.supp | 14
-b/doc/CHANGES-staging/README.md | 28
-b/doc/UPGRADE-staging/README.md | 31
-b/doc/appdocsxml.dtd | 2
-b/formats/format_g726.c | 16
-b/formats/msgsm.h | 4
-b/funcs/func_curl.c | 48
-b/funcs/func_env.c | 5
-b/funcs/func_jitterbuffer.c | 19
-b/funcs/func_odbc.c | 22
-b/funcs/func_pjsip_contact.c | 6
-b/funcs/func_talkdetect.c | 2
-b/include/asterisk/abstract_jb.h | 4
-b/include/asterisk/acl.h | 37
-b/include/asterisk/app.h | 195
-b/include/asterisk/ari.h | 2
-b/include/asterisk/astobj2.h | 5
-b/include/asterisk/audiohook.h | 2
-b/include/asterisk/autoconfig.h.in | 12
-b/include/asterisk/bridge.h | 9
-b/include/asterisk/calendar.h | 4
-b/include/asterisk/cel.h | 22
-b/include/asterisk/channel.h | 44
-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 | 20
-b/include/asterisk/dns_core.h | 22
-b/include/asterisk/dns_internal.h | 5
-b/include/asterisk/format_cache.h | 5
-b/include/asterisk/http_websocket.h | 14
-b/include/asterisk/iostream.h | 14
-b/include/asterisk/json.h | 16
-b/include/asterisk/max_forwards.h | 1
-b/include/asterisk/mixmonitor.h | 5
-b/include/asterisk/mwi.h | 242
-b/include/asterisk/netsock2.h | 42
-b/include/asterisk/options.h | 3
-b/include/asterisk/parking.h | 5
-b/include/asterisk/res_fax.h | 3
-b/include/asterisk/res_pjsip.h | 12
-b/include/asterisk/res_pjsip_presence_xml.h | 5
-b/include/asterisk/res_pjsip_pubsub.h | 23
-b/include/asterisk/res_pjsip_session.h | 2
-b/include/asterisk/rtp_engine.h | 129
-b/include/asterisk/sched.h | 16
-b/include/asterisk/serializer.h | 85
-b/include/asterisk/slin.h | 5
-b/include/asterisk/stasis.h | 46
-b/include/asterisk/stasis_bridges.h | 23
-b/include/asterisk/taskprocessor.h | 9
-b/include/asterisk/utils.h | 9
-b/main/abstract_jb.c | 178
-b/main/acl.c | 74
-b/main/aoc.c | 8
-b/main/app.c | 338
-b/main/asterisk.c | 18
-b/main/astmm.c | 23
-b/main/astobj2.c | 88
-b/main/astobj2_container.c | 24
-b/main/astobj2_global.c | 97
-b/main/astobj2_hash.c | 21
-b/main/astobj2_rbtree.c | 13
-b/main/audiohook.c | 11
-b/main/backtrace.c | 9
-b/main/bridge.c | 1
-b/main/ccss.c | 4
-b/main/cdr.c | 65
-b/main/cel.c | 82
-b/main/channel.c | 80
-b/main/codec_builtin.c | 8
-b/main/config.c | 33
-b/main/conversions.c | 39
-b/main/core_local.c | 6
-b/main/db.c | 3
-b/main/dns_core.c | 72
-b/main/event.c | 17
-b/main/features.c | 28
-b/main/file.c | 44
-b/main/format_cache.c | 8
-b/main/http.c | 56
-b/main/indications.c | 10
-b/main/iostream.c | 14
-b/main/json.c | 17
-b/main/loader.c | 149
-b/main/manager.c | 120
-b/main/manager_mwi.c | 1
-b/main/media_cache.c | 51
-b/main/message.c | 6
-b/main/mwi.c | 369
-b/main/options.c | 2
-b/main/pbx.c | 71
-b/main/pbx_variables.c | 23
-b/main/rtp_engine.c | 253
-b/main/say.c | 956
-b/main/sched.c | 30
-b/main/serializer.c | 189
-b/main/stasis.c | 479
-b/main/stasis_bridges.c | 36
-b/main/stasis_cache.c | 10
-b/main/stasis_channels.c | 9
-b/main/stream.c | 22
-b/main/taskprocessor.c | 219
-b/main/tcptls.c | 24
-b/main/translate.c | 13
-b/main/udptl.c | 18
-b/menuselect/Makefile | 8
-b/menuselect/autoconfig.h.in | 3
-b/menuselect/configure | 348
-b/menuselect/configure.ac | 12
-b/menuselect/example_menuselect-tree | 4
-b/menuselect/makeopts.in | 3
-b/menuselect/test/menuselect-tree | 4
-b/pbx/pbx_dundi.c | 135
-b/res/ael/pval.c | 30
-b/res/ari/ari_model_validators.c | 412
-b/res/ari/ari_model_validators.h | 55
-b/res/ari/config.c | 10
-b/res/ari/resource_channels.c | 216
-b/res/ari/resource_channels.h | 55
-b/res/ari/resource_events.c | 10
-b/res/parking/parking_bridge_features.c | 2
-b/res/parking/res_parking.h | 5
-b/res/res_agi.c | 20
-b/res/res_ari.c | 23
-b/res/res_ari_channels.c | 206
-b/res/res_ari_events.c | 2
-b/res/res_calendar_ews.c | 1
-b/res/res_calendar_exchange.c | 1
-b/res/res_calendar_icalendar.c | 1
-b/res/res_config_curl.c | 5
-b/res/res_config_pgsql.c | 2
-b/res/res_config_sqlite3.c | 6
-b/res/res_corosync.c | 2
-b/res/res_fax.c | 81
-b/res/res_http_websocket.c | 11
-b/res/res_musiconhold.c | 294
-b/res/res_mwi_devstate.c | 4
-b/res/res_mwi_external.c | 1
-b/res/res_phoneprov.c | 6
-b/res/res_pjsip.c | 94
-b/res/res_pjsip/config_global.c | 21
-b/res/res_pjsip/config_system.c | 2
-b/res/res_pjsip/config_transport.c | 17
-b/res/res_pjsip/location.c | 6
-b/res/res_pjsip/pjsip_configuration.c | 4
-b/res/res_pjsip/pjsip_message_filter.c | 53
-b/res/res_pjsip/pjsip_options.c | 47
-b/res/res_pjsip/pjsip_resolver.c | 4
-b/res/res_pjsip_config_wizard.c | 7
-b/res/res_pjsip_dialog_info_body_generator.c | 80
-b/res/res_pjsip_endpoint_identifier_ip.c | 102
-b/res/res_pjsip_mwi.c | 590
-b/res/res_pjsip_nat.c | 84
-b/res/res_pjsip_notify.c | 22
-b/res/res_pjsip_outbound_registration.c | 54
-b/res/res_pjsip_publish_asterisk.c | 2
-b/res/res_pjsip_pubsub.c | 107
-b/res/res_pjsip_refer.c | 5
-b/res/res_pjsip_registrar.c | 59
-b/res/res_pjsip_sdp_rtp.c | 29
-b/res/res_pjsip_session.c | 178
-b/res/res_pjsip_t38.c | 40
-b/res/res_pjsip_transport_websocket.c | 4
-b/res/res_realtime.c | 56
-b/res/res_remb_modifier.c | 67
-b/res/res_resolver_unbound.c | 6
-b/res/res_rtp_asterisk.c | 1214
-b/res/res_smdi.c | 1
-b/res/res_srtp.c | 33
-b/res/res_stasis.c | 45
-b/res/res_statsd.c | 35
-b/res/res_xmpp.c | 1
-b/res/stasis/app.c | 15
-b/res/stasis/command.c | 2
-b/res/stasis/control.c | 19
-b/rest-api-templates/res_ari_resource.c.mustache | 2
-b/rest-api/api-docs/channels.json | 329
-b/rest-api/resources.json | 2
-b/tests/CI/buildAsterisk.sh | 24
-b/tests/CI/findLibdir.sh | 14
-b/tests/CI/gateTestGroups.json | 7
-b/tests/CI/gates.jenkinsfile | 32
-b/tests/CI/installAsterisk.sh | 2
-b/tests/CI/periodic-dailyTestGroups.json | 6
-b/tests/CI/periodics-daily.jenkinsfile | 19
-b/tests/CI/ref_debug.jenkinsfile | 10
-b/tests/CI/unittests.jenkinsfile | 10
-b/tests/CI/universal-asterisk-nongerrit.jenkinsfile | 452
-b/tests/test_conversions.c | 48
-b/tests/test_data_buffer.c | 2
-b/tests/test_json.c | 18
-b/tests/test_res_rtp.c | 516
-b/tests/test_stasis_channels.c | 4
-b/tests/test_taskprocessor.c | 78
-b/third-party/jansson/Makefile | 7
-b/third-party/pjproject/Makefile | 18
-b/third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
-b/third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch | 53
-b/third-party/pjproject/patches/0030-ssl-regression-fix.patch | 105
-b/third-party/pjproject/patches/0031-transport-regression-fix.patch | 187
-doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt | 5
-doc/CHANGES-staging/rtp_ice_include_local_address.txt | 5
-doc/UPGRADE-staging/AMI-Originate.txt | 5
-third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch | 33
-third-party/pjproject/patches/0020-Fixed-2172-Avoid-double-reference-counter-decrements.patch | 42
-third-party/pjproject/patches/0031-Re-2191-transport-timer-cleanup.patch | 372
-third-party/pjproject/patches/0032-Re-2191-Fixed-crash-in-SIP-transport-destroy-due-to-.patch | 131
-316 files changed, 21717 insertions(+), 34013 deletions(-)</pre><br></html>
\ No newline at end of file
+++ /dev/null
- Release Summary
-
- asterisk-certified/16.8-cert1
-
- Date: 2020-04-30
-
- <asteriskteam@digium.com>
-
- ----------------------------------------------------------------------
-
- Table of Contents
-
- 1. Summary
- 2. Contributors
- 3. Closed Issues
- 4. Open Issues
- 5. Other Changes
- 6. Diffstat
-
- ----------------------------------------------------------------------
-
- Summary
-
- [Back to Top]
-
- This release is a point release of an existing major version. The changes
- included were made to address problems that have been identified in this
- release series, or are minor, backwards compatible new features or
- improvements. Users should be able to safely upgrade to this version if
- this release series is already in use. Users considering upgrading from a
- previous version are strongly encouraged to review the UPGRADE.txt
- document as well as the CHANGES document for information about upgrading
- to this release series.
-
- The data in this summary reflects changes that have been made since the
- previous release, asterisk-certified/16.3-cert1.
-
- ----------------------------------------------------------------------
-
- 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
- 53 Sean Bright 1 tests/test_utils.c. 16 Joshua C. Colp
- 51 George Joseph 1 Abhay Gupta 9 Ross Beer
- 25 Kevin Harwell 8 Kevin Harwell
- 25 Joshua Colp 6 Ross Beer
- 18 Asterisk Development 5 Frederic LE FOLL
- Team 5 George Joseph
- 14 Alexei Gradinari 4 Pascal Cadotte Michaud
- 13 Joshua C. Colp 4 Abhay Gupta
- 10 Ben Ford 4 Salah Ahmed
- 6 Corey Farrell 4 cmaj
- 6 Richard Mudgett 4 sungtae kim
- 5 Frederic LE FOLL 3 Dan Cropp
- 5 Jaco Kroon 3 Sean Bright
- 5 Tzafrir Cohen 3 Guido Falsi
- 4 sungtae kim 3 Dan Cropp
- 4 Abhay Gupta 3 nappsoft
- 4 Chris-Savinovich 2 Walter Doekes
- 3 Salah Ahmed 2 Joshua Elson
- 3 Pascal Cadotte Michaud 2 Bernhard Schmidt
- 3 Dan Cropp 2 Corey Farrell
- 3 Igor Goncharovsky 2 Stas Kobzar
- 3 Guido Falsi 2 Bernhard Schmidt
- 2 Rodrigo RamÃrez 2 Ruddy G
- Norambuena 2 Gregory Massel
- 2 Pirmin Walthert 2 Alexei Gradinari
- 2 Walter Doekes 2 Jonathan Harris
- 2 Torrey Searle 2 Torrey Searle
- 2 lvl 1 Oleksandr Natalenko
- 1 Matthew Fredrickson 1 Martin Tomec
- 1 Chris Savinovich 1 AvayaXAsterisk
- 1 Thomas Arimont (license 1 Jaco Kroon
- 5525) 1 Steven Wheeler
- 1 Nasir Iqbal 1 Byron Clark
- 1 Martin Tomec 1 candrews
- 1 Kevin Reeves 1 Jean Aunis - Prescom
- 1 Sebastian Kemper 1 Yoooooo Ha
- 1 Stas Kobzar 1 kevin@phoneburner.com
- 1 Francesco Castellano 1 Gil Richard
- 1 Jonathan Rose 1 Oleksandr Natalenko
- 1 Antoni Goldstein 1 Marian Piater
- 1 Morten Tryfoss 1 Michael Goryainov
- 1 Andrew Siplas 1 Niksa Baldun
- 1 Michael Goryainov 1 Alexander Traud
- 1 Jean Aunis 1 Mark
- 1 Leonid Fainshtein 1 Steven Wheeler
- 1 Lucas Mendes 1 Dirk Wendland
- 1 Michael Cargile 1 Bryan Nelson
- 1 Boris P. Korzun 1 Sam Banks
- 1 Florian Floimair 1 Sebastian Kemper
- 1 Holger Hans Peter 1 Speed Dial Dave
- Freyther 1 Richard Kenner
- 1 cmaj 1 Sébastien Duthil
- 1 Christoph Moench-Tegeder 1 Joshua C. Colp
- 1 Kirsty Tyerman 1 Sébastien Duthil
- 1 snuffy 1 Aheliotech
- 1 Asterisk Team 1 Jim Van Meggelen
- 1 Alexander Anikin 1 Robert Sutton
- 1 Michael Cargile
- 1 Kevin Flyn
- 1 Janu
- 1 Frank Matano
- 1 pasandev
- 1 Cédric Bassaget
- 1 Kevin Flyn
- 1 Dan Jenkins
- 1 Luke-Jr
- 1 Robert Sutton
- 1 Jeremiah Gadd
- 1 Michael
- 1 Boris P. Korzun
- 1 Kilburn
- 1 Alexander Traud
- 1 Joeran Vinzens
- 1 Dennis
- 1 test011
- 1 Joeran Vinzens
- 1 Jim Van Meggelen
- 1 Kirill Katsnelson
- 1 Kirsty Tyerman
- 1 Lucas Mendes
- 1 Timothy Vanderaerden
- 1 Janu
- 1 Florian Floimair
- 1 Michael Maier
- 1 Daniel
- 1 Dan Jenkins
- 1 Robin Leffmann
- 1 Mitch Claborn
- 1 Antoni Goldstein
- 1 David Lee
- 1 Dmitry Shubin
- 1 klaus3000
- 1 Maciej Michno
- 1 Dmitry Shubin
- 1 Gil Richard
- 1 Kevin Reeves
- 1 Niklas Larsson
- 1 Dmitry Svyatogorov
- 1 Jean-Denis Girard
- 1 Christoph
- Moench-Tegeder
- 1 Maciej Michno
- 1 the CC variable,
- instead of
- unconditionally
- 1 Stas Kobzar
- 1 Francesco Castellano
- 1 Cedric BASSAGET
- 1 Ted G
- 1 Frank Matano
- 1 David M. Lee
- 1 vijay kumar
- 1 Niklas Larsson
- 1 Andrey V. T.
- 1 Francois Blackburn
- 1 Juan Martin
- 1 Richard Kenner
- 1 Abhay Gupta
- 1 Ian Jones
- 1 Jean-Denis Girard
- 1 lvl
- 1 Ted G
- 1 Marin Odrljin
- 1 Morten Tryfoss
- 1 Andrew Siplas
- 1 Vyrva Igor
- 1 Jonas Swiatek
- 1 Eliel Sardañons
- 1 AvayaXAsterisk
- 1 Dirk Wendland
- 1 Luke-Jr
- 1 abelbeck
- 1 Jonathan Harris
- 1 Nasir Iqbal
- 1 Chris Savinovich
- 1 Kirill Katsnelson
- 1 Eliel Sardañons
- 1 Sean Bright
- 1 Kirsty Tyerman
- 1 Cyril Ramière
- 1 Jørgen H
- 1 Niksa Baldun
- 1 dennis
-
- ----------------------------------------------------------------------
-
- 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.
- * [8cdaa93e65] Ben Ford -- chan_sip.c: Prevent address change on
- unauthenticated SIP request.
-
- Category: Channels/chan_sip/Interoperability
-
- ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE
- Reported by: Francesco Castellano
- * [6c59df17a5] Francesco Castellano -- chan_sip: Handle invalid SDP
- answer to T.38 re-invite
-
- Category: Core/ManagerInterface
-
- ASTERISK-28580: Bypass SYSTEM write permission in manager action allows
- system commands execution
- Reported by: Eliel Sardañons
- * [7574be5110] George Joseph -- manager.c: Prevent the Originate action
- from running the Originate app
-
- Category: Resources/res_pjsip_messaging
-
- ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes
- crash
- Reported by: Gil Richard
- * [2126dc3021] George Joseph -- res_pjsip_messaging: Check for body in
- in-dialog message
-
- Category: Resources/res_pjsip_t38
-
- ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream
- causes crash
- Reported by: Alexei Gradinari
- * [965df3c228] 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
- * [8894a56452] lvl -- app_senddtmf: Add receive mode to AMI Action
- PlayDTMF
-
- Category: Core/Jitterbuffer
-
- ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
- Reported by: Joshua C. Colp
- * [6647be69ac] 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
- * [f69da94fab] Sean Bright -- func_curl: Add 'followlocation' option to
- CURLOPT()
- ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
- Reported by: Martin Tomec
- * [37dcdd485a] Martin Tomec -- func_curl.c: Support custom http headers
-
- Category: Resources/res_ari_channels
-
- ASTERISK-28320: Added ARI resource
- /ari/channels/{channelid}/rtp_statistics
- Reported by: sungtae kim
- * [bbc13b1f1f] sungtae kim -- res/res_ari: Added ARI resource
- /ari/channels/{channelId}/rtp_statistics
-
- Category: Resources/res_musiconhold
-
- ASTERISK-17808: [patch] Unregister a realtime moh class
- Reported by: Byron Clark
- * [b478f46d59] sungtae kim -- res_musiconhold: Added unregister realtime
- moh class
-
- Category: Resources/res_pjsip
-
- ASTERISK-28375: res_pjsip: New configuration setting to allow disabling
- norefersub
- Reported by: Dan Cropp
- * [eca8c440d2] Dan Cropp -- res_pjsip: Added a norefersub configuration
- setting
-
- 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
- * [f8b0c2c933] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add
- port matching support
-
- Category: Resources/res_pjsip_refer
-
- ASTERISK-28375: res_pjsip: New configuration setting to allow disabling
- norefersub
- Reported by: Dan Cropp
- * [eca8c440d2] Dan Cropp -- res_pjsip: Added a norefersub configuration
- setting
-
- Category: pjproject/pjsip
-
- ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup
- From header URI domain
- Reported by: Stas Kobzar
- * [fb984eda40] 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
- * [7db5f5df6a] Kevin Harwell -- res_pjsip_mwi: add better handling of
- solicited vs unsolicited subscriptions
-
- Category: Addons/chan_ooh323
-
- ASTERISK-28348: Failed to initialize OOH323 endpoint-OOH323 Disabled
- Reported by: Dmitry Shubin
- * [eec16b8e99] Alexander Anikin -- chan_ooh323: fix h323 log file path
-
- Category: Applications/app_amd
-
- ASTERISK-28608: app_amd: Use time calculation to calculate timeout
- Reported by: Michael Cargile
- * [cf5b7f3a0b] Michael Cargile -- app_amd: Fixed timeout issue
- ASTERISK-28419: app_amd: Does not work with silence suppression
- Reported by: Nasir Iqbal
- * [52a3d4a761] Nasir Iqbal -- app_amd: issue with silence suppression
- fixed
- ASTERISK-28143: app_amd: Infinite loop on silent calls
- Reported by: Abhay Gupta
- * [1d214a3623] Abhay Gupta -- app_amd: Fix infinite loop on silent calls
-
- Category: Applications/app_chanisavail
-
- ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to
- deactivate CDR.
- Reported by: Frederic LE FOLL
- * [aa06c6ea29] 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
- * [c8cf3ad389] Frederic LE FOLL -- ChanIsAvail() generates a CDR when
- unanswered=yes in cdr.conf.
-
- Category: Applications/app_confbridge
-
- ASTERISK-28790: Crash during conference call using confbridge and video
- Reported by: Pascal Cadotte Michaud
- * [3d22ecb1e2] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient
- space for worst case NACK.
-
- Category: Applications/app_meetme
-
- ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on
- 17.0.0
- Reported by: George Joseph
- * [7167fd6d46] Joshua C. Colp -- configure: Add check for MySQL client
- bool and my_bool type usage.
- * [be4c6f3f35] George Joseph -- cdr_mysql: Fix missing use of 'my_bool'
- with MySql >= 8.0.1
-
- Category: Applications/app_queue
-
- ASTERISK-28349: Pause reason not reported in QueueMember AMI event
- Reported by: Niksa Baldun
- * [5fded77e7f] Sean Bright -- app_queue: Deprecate the
- QueueMemberPause.Reason field
- ASTERISK-28644: Stale comment in app_queue about ring_entry exception
- Reported by: Walter Doekes
- * [161e762742] Walter Doekes -- app_queue: Fix old confusing comment
- about when the members are called
-
- Category: Applications/app_record
-
- ASTERISK-28682: app_record: Lack of `beep` audio file causes application
- to return error and hangup
- Reported by: Corey Farrell
- * [0c07a7ee00] Corey Farrell -- app_record: Do not hang up if beep audio
- is missing
-
- Category: Applications/app_transfer
-
- ASTERISK-26968: chan_pjsip: Transfer() does not result in TRANSFERSTATUS
- reflecting SIP response to transfer
- Reported by: Dan Cropp
- * [f4896703b9] Dan Cropp -- chan_pjsip: Transmit REFER waits for the
- REFER result setting TRANSFERSTATUS
-
- Category: Applications/app_voicemail
-
- ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage
- enabled and realtime voicemail_data is used
- Reported by: Stas Kobzar
- * [293600724d] 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
- * [e379fe48e1] Sean Bright -- app_voicemail: Set globals to default
- values when voicemail.conf missing
- ASTERISK-27935: app_voicemail: emailbody per user can't contain commas
- Reported by: Sébastien Duthil
- * [d0a8334e4f] Sean Bright -- app_voicemail: Don't split mailbox options
- on comma
-
- Category: Applications/app_voicemail/IMAP
-
- ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because
- not checking mailstream
- Reported by: Alexei Gradinari
- * [ff180a5bfc] 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
- * [293600724d] Sean Bright -- app_voicemail: Prevent crash when saving
- message with realtime voicemail
-
- Category: Bridges/bridge_native_rtp
-
- ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
- check failure when negociated ptime is not default ptime.
- Reported by: Frederic LE FOLL
- * [31173f6586] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
- directmedia for ptime other than default ptime.
-
- Category: Bridges/bridge_softmix
-
- ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix
- bridge
- Reported by: Kevin Harwell
- * [3891a953cf] 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
- * [1b452ebb51] 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
- * [aa06c6ea29] 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
- * [495bc77a6a] 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
- * [6c54bd704e] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql
- res_config_pgsql: compatibility with PostgreSQL 12
- ASTERISK-28435: cdr_pgsql: Unix socket doesn't work
- Reported by: Dmitry Svyatogorov
- * [c2621aa190] Chris-Savinovich -- cdr_pgsql: fix error in connection
- string
-
- 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
- * [9895e94dba] 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
- * [faf353e931] 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
- * [448b8c9bc2] 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
- * [c6b17b5212] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource)
- when a PRI channel hangs up
- ASTERISK-28457: [patch] Fix crash in chan_dahdi on 32-bit systems caused
- by ASTERISK-28317
- Reported by: abelbeck
- * [da1db4f842] Chris-Savinovich -- chan_dahdi.c: crash in chan_dahdi
- ASTERISK-28427: new mwi.h include missing from some dahdi source files,
- causes build failure
- Reported by: Guido Falsi
- * [86fb72c4d0] Guido Falsi -- chan_dahdi: add missing include.
-
- Category: Channels/chan_local
-
- ASTERISK-28399: channel.c: Exceptionally long queue length queuing
- Reported by: Abhay Gupta
- * [9a0fa51443] Abhay Gupta -- stasis: Hangup channel for Local channel
- No such extension error
-
- Category: Channels/chan_pjsip
-
- ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group
- endpoint/call_group
- Reported by: Jean-Denis Girard
- * [992dcdf780] 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
- * [63b8664bfa] George Joseph -- res_pjsip_nat: Restore original contact
- for REGISTER responses
- ASTERISK-28578: race condition on pjsip channelstats command
- Reported by: Salah Ahmed
- * [c73aaa2760] Salah Ahmed -- Crash during "pjsip show channelstats"
- execution
- ASTERISK-28561: Asterisk Deadlocks
- Reported by: Aheliotech
- * [a66848c92f] 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
- * [19565e0b90] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF
- on hungup channel
- ASTERISK-28538: chan_pjsip: Deadlock on fax detection
- Reported by: Joshua C. Colp
- * [49e1346185] Joshua Colp -- chan_pjsip: Relock correct channel during
- "fax" redirect.
- ASTERISK-26968: chan_pjsip: Transfer() does not result in TRANSFERSTATUS
- reflecting SIP response to transfer
- Reported by: Dan Cropp
- * [f4896703b9] Dan Cropp -- chan_pjsip: Transmit REFER waits for the
- REFER result setting TRANSFERSTATUS
- ASTERISK-28444: chan_pjsip: Peer IP for SSL handshake errors not logged
- Reported by: Bernhard Schmidt
- * [2db5173b88] George Joseph -- pjproject_bundled: Add peer information
- to most SSL/TLS errors
- ASTERISK-25371: Crash in hangup at chan_pjsip.c:1749 when Asterisk
- attempts to generate hangup event
- Reported by: Abhay Gupta
- * [72f26aa8eb] Abhay Gupta -- chan_pjsip.c: Check for channel and
- session to not be NULL in hangup
- ASTERISK-27994: PJSIP: Early media ringback not indicated after Progress()
- Reported by: Gregory Massel
- * [de82bdd746] Alexei Gradinari -- pjsip: replace 180 by 183 if SDP
- negotiation has completed
- ASTERISK-28379: pjsip: show channelstats incorrect information output
- Reported by: Vyrva Igor
- * [ece29db9bd] Joshua Colp -- res_rtp_asterisk: Fix sequence number
- cycling and packet loss count.
- ASTERISK-28371: chan_pjsip: DTMF Mode auto_info fallback lead to both
- inband and info
- Reported by: Salah Ahmed
- * [7e5b4b8616] Salah Ahmed -- chan_pjsip: DTMF Mode auto_info fallback
- lead to both inband and info
-
- Category: Channels/chan_sip/General
-
- ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
- Reported by: Jean Aunis - Prescom
- * [82a870c8c7] Jean Aunis -- chan_sip: voice frames are no longer
- transmitted after emitting a COLP
- ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
- Reported by: Jaco Kroon
- * [055737d645] Jaco Kroon -- chan_sip: in case of tcp/tls, be less
- annoying about tx errors.
- ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility
- check failure when negociated ptime is not default ptime.
- Reported by: Frederic LE FOLL
- * [31173f6586] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no
- directmedia for ptime other than default ptime.
- ASTERISK-28282: AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
- chan_sip)
- Reported by: Walter Doekes
- * [64d25d36fb] Walter Doekes -- sched: Don't allow ast_sched_del to
- deadlock ast_sched_runq from same thread
- ASTERISK-28362: strtok_r() makes gcc compile warning
- Reported by: sungtae kim
- * [8961d9ca8b] Ben Ford -- build: Fix compiler warnings/errors.
-
- Category: Channels/chan_sip/Messaging
-
- ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears
- empty in the dialplan
- Reported by: Frank Matano
- * [31027f33db] Sean Bright -- chan_sip.c: Stop handling continuation
- lines after reading headers
-
- Category: Channels/chan_sip/TCP-TLS
-
- ASTERISK-26006: Show offending IP for TLS setup failures in logs
- Reported by: Oleksandr Natalenko
- * [0dc61e41fa] George Joseph -- tcptls.c: Add peer hostname and port to
- some error messages
-
- Category: Channels/chan_sip/Transfers
-
- ASTERISK-28677: CDR billsec is always 0 for transferred calls
- Reported by: Maciej Michno
- * [1b452ebb51] George Joseph -- cdr.c: Set event time on party b when
- leaving a parking bridge
-
- Category: Channels/chan_unistim
-
- ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at
- end of a struct
- Reported by: Alexander Traud
- * [92261d60c8] 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
- * [bf527810ef] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is
- defined when necessary
- * [cdbb9800e3] 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
- * [efecc9d139] 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
- * [a78758d0a2] Sean Bright -- config.c: Skip UTF-8 BOMs if present when
- reading config files
-
- Category: Contrib/General
-
- ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed
- to due to invalid syntax
- Reported by: Richard Kenner
- * [b9b50774f5] snuffy -- contrib/valgrind: Fix use of frame-level
- suppression
- ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
- Reported by: Pascal Cadotte Michaud
- * [b8e635916f] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid
- typo
-
- Category: Core/BuildSystem
-
- ASTERISK-28487: compile menuselect on gentoo
- Reported by: Kilburn
- * [8399211eaf] Sean Bright -- menuselect: Fix curses build on Gentoo
- Linux
- ASTERISK-28392: The no-partial-inlining flag isn't passed to the bundled
- pjproject or jansson builds
- Reported by: George Joseph
- * [543d487746] George Joseph -- build: Pass --fno-partial-inlining to
- third-party when appropriate
- ASTERISK-28374: latest asterisk unconditionally launch gcc --version, even
- if the compiler is different
- Reported by: Guido Falsi
- * [4dcfa8d127] Guido Falsi -- core/buildsystem: check the actual
- compiler being version
-
- Category: Core/Channels
-
- ASTERISK-28795: channel: write to a stream on multi-frame writes
- Reported by: Kevin Harwell
- * [3297df5a61] 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
- * [2691ee7e10] Joshua Colp -- AST-2019-005 - translate: Don't assume all
- frames will have a src.
-
- Category: Core/Configuration
-
- ASTERISK-23756: setvar directive when used in template and a child of said
- template, results in duplicate variable names
- Reported by: Michael Goryainov
- * [518b6bfb5c] Michael Goryainov -- channels: Allow updating variable
- value
-
- Category: Core/General
-
- ASTERISK-28498: cel / cdr: Event times may be incorrect
- Reported by: Joshua C. Colp
- * [6350f4e278] Joshua Colp -- cdr / cel: Use event time at event
- creation instead of processing.
-
- Category: Core/Logging
-
- ASTERISK-26006: Show offending IP for TLS setup failures in logs
- Reported by: Oleksandr Natalenko
- * [0dc61e41fa] George Joseph -- tcptls.c: Add peer hostname and port to
- some error messages
-
- Category: Core/RTP
-
- ASTERISK-28480: json integer overflow in ssrc and timestamp
- Reported by: Salah Ahmed
- * [6bb14150c4] Kevin Harwell -- various modules: json integer overflow
-
- Category: Core/Stasis
-
- ASTERISK-28553: stasis.c: Crash during unload
- Reported by: Kevin Harwell
- * [ce1e0714ba] Joshua Colp -- stasis: Pass bumped topic_all reference to
- proxy_dtor.
- ASTERISK-28353: stasis: Crash at shutdown when statistics enabled
- Reported by: Joshua C. Colp
- * [8d35a30a3f] Ben Ford -- stasis: Fix crash at shutdown.
-
- Category: Core/Streams
-
- ASTERISK-28625: Playback of local files impacted by large media cache
- Reported by: Kevin Reeves
- * [e013f502b1] 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
- * [83390327b2] Torrey Searle -- main/udptl.c: correctly handle udptl
- sequence wrap around
-
- Category: Documentation
-
- ASTERISK-24484: Update documentation for statsd module - usage
- requirements unclear
- Reported by: Dan Jenkins
- * [04c81f9748] 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
- * [8d87fef5a1] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
- support for hostnames
- ASTERISK-28507: Wiki docs missing for MessageWaiting
- Reported by: David M. Lee
- * [4cf32f2578] 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
- * [2d2b28bfa4] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing
- argument documentation
- * [174e6426aa] 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
- * [e39ddb1cb1] Sean Bright -- func_env: Prevent FILE() from reading
- garbage at end-of-file
-
- Category: Functions/func_odbc
-
- ASTERISK-28497: func_odbc: truncating Unicode string on readsql
- Reported by: Boris P. Korzun
- * [e54299cd3e] Boris P. Korzun -- func_odbc: acf_odbc_read() and
- cli_odbc_read() unicode support
-
- Category: General
-
- ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
- Reported by: Ted G
- * [8af0dea0c7] 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
- * [e627db7928] 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
- * [bd96a0b79d] 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
- * [d4766a82a2] Kevin Harwell -- srtp: Fix possible race condition, and
- add NULL checks
- ASTERISK-28412: GCC 9 catches more string formatting issues
- Reported by: George Joseph
- * [e7734476c6] George Joseph -- Fixes for GCC 9
- ASTERISK-28319: musl: Crash on startup when loading modules
- Reported by: Sebastian Kemper
- * [8ec4de7501] Sebastian Kemper -- loader: support for permanent
- dlopen()
-
- Category: PBX/General
-
- ASTERISK-28695: core: minmemfree watermark uses free RAM, not available
- RAM
- Reported by: Kevin Flyn
- * [f5a1e8b04d] 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
- * [64692a3c72] George Joseph -- sig_pri: Fix deadlock caused by
- sig_pri_queue_hangup
- ASTERISK-20182: Parsing a label beginning with a numeric character in all
- Goto/GotoIf/GotoIfTime application causes unexpected behavior
- Reported by: Janu
- * [f91262272e] Sean Bright -- pbx.c: Properly parse labels with leading
- digits
-
- Category: PBX/pbx_ael
-
- ASTERISK-17799: AEL reload causes loss of control in a macro
- Reported by: Kirill Katsnelson
- * [116dc9c9b3] Sean Bright -- res_ael: Create consistent label names
- across reloads
- ASTERISK-18593: AEL for loops use Macro app and pipe delimiter
- Reported by: Luke-Jr
- * [ea3109beaa] Sean Bright -- res_ael: Use Gosub in for loop expressions
- ASTERISK-14939: AEL parsers does not find existing label
- Reported by: klaus3000
- * [71c7864d1d] Sean Bright -- res_ael: Fix pattern matching against
- literal '+'
-
- Category: PBX/pbx_config
-
- ASTERISK-28534: Segmentation fault when there is no priority for an
- extension
- Reported by: Timothy Vanderaerden
- * [0514559005] Sean Bright -- pbx: Prevent Realtime switch crash on
- invalid priority
-
- Category: Resources/res_ari
-
- ASTERISK-28679: stasis application is destroyed after its creation
- Reported by: Francois Blackburn
- * [1627e8eddc] Kevin Harwell -- res_stasis: trigger cleanup after update
- ASTERISK-28585: ari/resource_events: Crash in event session cleanup
- Reported by: Kevin Harwell
- * [6e22e1213e] Joshua Colp -- res_ari_events: Add module reference when
- a WebSocket is open.
- ASTERISK-26718: ARI: Bridge destroying doesn't work as expected
- Reported by: Marin Odrljin
- * [f599ebd29e] Holger Hans Peter Freyther -- stasis: Call callbacks when
- imparting fails
-
- Category: Resources/res_calendar_exchange
-
- ASTERISK-28572: Memory leaks in res_calendar_exchange and
- res_calendar_icalendar
- Reported by: Yoooooo Ha
- * [ccaf735d1f] 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
- * [ccaf735d1f] Sean Bright -- res_calendar: Resolve memory leak on
- calendar destruction
-
- Category: Resources/res_config_sqlite3
-
- ASTERISK-28477: Crash when not specifying "dbfile" in
- res_config_sqlite3.conf
- Reported by: Dennis
- * [28654308ef] Sean Bright -- res_config_sqlite3: Only join threads that
- we started
- ASTERISK-28478: Crash performing "core reload" with modified
- res_config_sqlite3.conf
- Reported by: Dennis
- * [28654308ef] Sean Bright -- res_config_sqlite3: Only join threads that
- we started
-
- Category: Resources/res_fax
-
- ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config
- option
- Reported by: Kevin Harwell
- * [d17bbcb9f1] Kevin Harwell -- res_fax: wrap v21 detected Asterisk
- initiated negotiation with config option
- ASTERISK-27981: res_fax: Fax session leak with fax gatewaying
- Reported by: pasandev
- * [e0a574253e] Alexei Gradinari -- res_fax: fix segfault on inactive
- "reserved" fax session
-
- Category: Resources/res_http_websocket
-
- ASTERISK-28562: SIP WSS message not processed until next frame arrives
- Reported by: Robert Sutton
- * [47ba42f4a0] Sean Bright -- websocket: Consider pending SSL data when
- waiting for socket input
-
- Category: Resources/res_indications
-
- ASTERISK-28391: res_indications: Crash requesting autocomplete on
- indications cli command
- Reported by: Lucas Mendes
- * [daed593cfa] Lucas Mendes -- res_indications: Fix indications remove
- command autocomplete
-
- Category: Resources/res_parking
-
- ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the
- same
- Reported by: Ross Beer
- * [c91b28c72d] 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
- * [b9bbf39449] Joshua Colp -- parking: Fix case where we can't get the
- parker.
- * [e7320bbbf0] Joshua Colp -- parking: Use channel snapshot instead of
- channel.
-
- Category: Resources/res_pjsip
-
- ASTERISK-28790: Crash during conference call using confbridge and video
- Reported by: Pascal Cadotte Michaud
- * [3d22ecb1e2] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient
- space for worst case NACK.
- ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR
- points to a not existent AOR
- Reported by: Ross Beer
- * [68ce999351] 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
- * [1f10ca76da] Sean Bright -- res_pjsip_transport_websocket: Don't put
- brackets around local_name if IPv6
- ASTERISK-28521: pjsip: Memory Leak
- Reported by: Mark
- * [789c51ac8a] 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
- * [c2b135729c] Joshua Colp -- res_pjsip: Fix multiple of the same
- contact in "pjsip show contacts".
-
- Category: Resources/res_pjsip_endpoint_identifier_ip
-
- ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for
- hostnames
- Reported by: Joshua C. Colp
- * [8d87fef5a1] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document
- support for hostnames
-
- Category: Resources/res_pjsip_mwi
-
- ASTERISK-28575: MWI Send Notify Crash on 16.6
- Reported by: Joshua Elson
- * [17e71b6abe] 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
- * [3f12cd7711] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object
- for mwi containers
-
- Category: Resources/res_pjsip_notify
-
- ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present
- instead of just one
- Reported by: AvayaXAsterisk
- * [0a56edca4d] 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
- * [bb19e7feb5] George Joseph -- res_pjsip_outbound_registration: Fix SRV
- failover on timeout
- ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
- Reported by: Kevin Harwell
- * [3cd57aaff2] Kevin Harwell -- res_pjsip_outbound_registration: add
- support for SRV failover
- ASTERISK-28521: pjsip: Memory Leak
- Reported by: Mark
- * [789c51ac8a] 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
- * [41cd1ff454] 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
- * [939e18d63e] 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
- * [8318b05f25] Joshua C. Colp -- res_pjsip_pubsub: Add ability to
- persist generator state information.
-
- Category: Resources/res_pjsip_registrar
-
- ASTERISK-28402: res_pjsip_registrar: SEGV in registrar_find_contact
- Reported by: Ross Beer
- * [5002169d6a] George Joseph -- res_pjsip: Check return from
- pjsip_parse_uri calls
-
- Category: Resources/res_pjsip_sdp_rtp
-
- ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media
- stream if codecs create additional streams and offer does not have them
- Reported by: nappsoft
- * [186c4e9b36] Joshua C. Colp -- res_pjsip_session: Set stream state on
- created streams for incoming SDP.
- ASTERISK-28458: res_pjsip_sdp_rtp: Remove unused variable
- Reported by: Michael Maier
- * [83aba363fe] Kevin Harwell -- res_pjsip_sdp_rtp: Remove unused
- variable
-
- Category: Resources/res_pjsip_session
-
- ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have
- reflected state
- Reported by: Joshua C. Colp
- * [aa04c3f49b] 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
- * [d515dea9c6] 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
- * [186c4e9b36] 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
- * [fbc80db350] 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
- * [19565e0b90] 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
- * [efef44985d] 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
- * [fbe18165d5] Sean Bright -- res_realtime: Fix 'realtime update2'
- argument handling
-
- Category: Resources/res_rtp_asterisk
-
- 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
- * [22bc8a7168] 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
- * [fef8a04aad] Pirmin Walthert -- res_rtp_asterisk: Free payload when
- error on insertion to data buffer
- ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling
- Reported by: Joshua C. Colp
- * [4458b7a52a] Joshua C. Colp -- res_rtp_asterisk: Improve video
- performance in certain networks.
- ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete
- dtls/srtp setup
- Reported by: Kevin Harwell
- * [43ca35b831] 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
- * [eb54e381fd] Joshua Colp -- res_rtp_asterisk: Always return provided
- DTLS packet length.
- ASTERISK-28018: IP Fragmentation happening instead of DTLS fragmentation
- on handshake server hello certificate
- Reported by: vijay kumar
- * [82789aafd6] Joshua Colp -- res_rtp_asterisk: Add support for DTLS
- packet fragmentation.
- ASTERISK-28421: Wrong type used for timestamp in res_rtp_asterisk
- Reported by: Morten Tryfoss
- * [9351aa3f0e] Morten Tryfoss -- res_rtp_asterisk: timestamp should be
- unsigned instead of signed int
- ASTERISK-28255: res_rtp_asterisk: REMB RTCP packet sending may be
- incorrect
- Reported by: Joshua C. Colp
- * [94eeba6147] Kevin Harwell -- bridge_softmix: use a float type to
- store the internal REMB bitrate
-
- Category: Resources/res_stasis
-
- ASTERISK-28423: ARI causes STASIS Deadlock
- Reported by: Ross Beer
- * [42c51263b9] Kevin Harwell -- stasis/app: don't lock an app before a
- call to send
- * [40c49ec64f] George Joseph -- stasis: Don't hold app_registry and
- session locks unnecessarily
- ASTERISK-28633: stasis bridge topic leak
- Reported by: Joeran Vinzens
- * [dd82ebecd3] George Joseph -- stasis.c: Use correct topic name in
- stasis_topic_pool_delete_topic
- ASTERISK-27756: bridge: Failure to impart a channel results in bad data
- causing crash
- Reported by: Abhay Gupta
- * [39c5188bec] Abhay Gupta -- stasis: Only place stasis created and
- dialed channels into dial bridge.
- ASTERISK-26718: ARI: Bridge destroying doesn't work as expected
- Reported by: Marin Odrljin
- * [f599ebd29e] Holger Hans Peter Freyther -- stasis: Call callbacks when
- imparting fails
-
- Category: Resources/res_statsd
-
- ASTERISK-24484: Update documentation for statsd module - usage
- requirements unclear
- Reported by: Dan Jenkins
- * [04c81f9748] Sean Bright -- res_statsd: Document that res_statsd does
- nothing on its own
-
- Category: Tests/NewFeature
-
- ASTERISK-17695: 1.8.3.2 extenpatternmatchnew=yes cannot find extensions
- with '-' in them
- Reported by: test011
- * [9b7a64cbf0] Sean Bright -- pbx.c: Ignore dashes in extensions when
- using extenpatternmatchnew
-
- Category: Utilities/aelparse
-
- ASTERISK-18593: AEL for loops use Macro app and pipe delimiter
- Reported by: Luke-Jr
- * [ea3109beaa] Sean Bright -- res_ael: Use Gosub in for loop expressions
-
- Category: Utilities/conf2ael
-
- ASTERISK-18593: AEL for loops use Macro app and pipe delimiter
- Reported by: Luke-Jr
- * [ea3109beaa] Sean Bright -- res_ael: Use Gosub in for loop expressions
-
- Category: pjproject/pjsip
-
- ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
- Reported by: Niklas Larsson
- * [cb6e0d36ae] 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
- * [c8cc530726] Dan Cropp -- pjproject: Configurable setting for cnonce
- to include hyphens or not
-
- Improvement
-
- Category: Applications/app_confbridge
-
- ASTERISK-28658: app_confbridge: Add support for setting maximum sample
- rate
- Reported by: Joshua C. Colp
- * [5622df0a94] Joshua C. Colp -- confbridge: Add support for specifying
- maximum sample rate.
- ASTERISK-28401: app_confbridge: Add *_all remb behavior variants
- Reported by: Joshua C. Colp
- * [d861ebdca8] Joshua Colp -- app_confbridge: Add "all" variants of REMB
- behavior.
-
- Category: Applications/app_dial
-
- ASTERISK-28363: Millisecond-resolution call stats including PDD in channel
- variables
- Reported by: Antoni Goldstein
- * [d6b37e2926] Antoni Goldstein -- app_dial.c: RINGTIME, PROGRESSTIME
- and ms resolution dial timings
-
- 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
- * [68855f6a7b] Sean Bright -- Revert "app_voicemail: Cleanup stale lock
- files on module load"
- ASTERISK-20207: Asterisk should clear out any .lock files in the voice
- mail directory on startup.
- Reported by: Steven Wheeler
- * [34b9b65098] Sean Bright -- 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
- * [b41a8fd0f3] 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
- * [5fccf6d1c6] 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
- * [5fccf6d1c6] 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
- * [5fccf6d1c6] 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
- * [5622df0a94] Joshua C. Colp -- confbridge: Add support for specifying
- maximum sample rate.
- ASTERISK-28401: app_confbridge: Add *_all remb behavior variants
- Reported by: Joshua C. Colp
- * [d861ebdca8] Joshua Colp -- app_confbridge: Add "all" variants of REMB
- behavior.
-
- Category: Channels/chan_pjsip
-
- ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
- Reported by: cmaj
- * [a7692ce2f4] Richard Mudgett -- app_chanisavail.c: Simplify dialplan
- using ChanIsAvail.
- * [144b774b85] Richard Mudgett -- app_dial.c: Simplify dialplan using
- Dial.
- * [2780be334d] Richard Mudgett -- app_page.c: Simplify dialplan using
- Page.
-
- Category: Core/BuildSystem
-
- ASTERISK-28111: build: CHANGES/UPGRADE are irritating to work with.
- Reported by: Corey Farrell
- * [ef404fef91] Ben Ford -- build: Revise CHANGES and UPGRADE.txt
- handling.
-
- Category: Core/CodecInterface
-
- ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
- Reported by: Florian Floimair
- * [f85631cf82] Florian Floimair -- core: Add H.265/HEVC passthrough
- support
-
- Category: Core/HTTP
-
- ASTERISK-28710: Should be able to disable the /httpstatus URI in the
- built-in HTTP server
- Reported by: Sean Bright
- * [a2a4e1026c] Sean Bright -- http: Add ability to disable /httpstatus
- URI
-
- Category: Core/Streams
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5fccf6d1c6] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
-
- Category: Documentation
-
- ASTERISK-28673: GET FULL VARIABLE documentation clarification
- Reported by: Jonathan Harris
- * [60fd1322d7] Sean Bright -- res_agi: Improve GET FULL VARIABLE
- documentation
- ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
- Reported by: Sam Banks
- * [2666a5e111] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling
- correetions.
-
- Category: PBX/General
-
- ASTERISK-28264: Added topic_all container
- Reported by: sungtae kim
- * [5815597a21] sungtae kim -- stasis.c: Added topic_all container
-
- Category: PBX/pbx_dundi
-
- ASTERISK-28234: pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
- Reported by: Kirsty Tyerman
- * [a1c84709b8] Kirsty Tyerman -- pbx_dundi: added IPv4/IPv6 dual bind
- support to DUNDi
-
- Category: Resources/res_ari_channels
-
- ASTERISK-28343: Added app_name, app_data to channel type
- Reported by: sungtae kim
- * [d5a318f148] sungtae kim -- main/json.c: Added app_name, app_data to
- channel type
-
- Category: Resources/res_pjsip
-
- ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold
- re-invites
- Reported by: Torrey Searle
- * [9a933c3adc] Torrey Searle -- channel/chan_pjsip: add dialplan
- function for music on hold
-
- Category: Resources/res_pjsip_outbound_registration
-
- ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
- Reported by: Daniel
- * [95bc698b85] Joshua Colp -- res_pjsip_outbound_registration: Extend
- documentation for "max_retries".
-
- Category: Resources/res_pjsip_sdp_rtp
-
- ASTERISK-28733: stream: Add support for adding/removing streams during
- SFU/calls
- Reported by: Joshua C. Colp
- * [5fccf6d1c6] Joshua C. Colp -- bridging: Add better support for
- adding/removing streams.
- ASTERISK-28400: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
- transport-cc
- Reported by: Joshua C. Colp
- * [5023f02b2d] Joshua Colp -- rtp: Add support for transport-cc in
- receiver direction.
-
- Category: Resources/res_pjsip_session
-
- ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add
- video
- Reported by: Joshua C. Colp
- * [bdf4d159fd] Joshua C. Colp -- res_pjsip_session: Apply intention
- behind requested formats.
-
- Category: Resources/res_rtp_asterisk
-
- ASTERISK-28400: res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
- transport-cc
- Reported by: Joshua C. Colp
- * [5023f02b2d] Joshua Colp -- rtp: Add support for transport-cc in
- receiver direction.
-
- ----------------------------------------------------------------------
-
- 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: Resources/res_pjsip
-
- ASTERISK-28161: Removal of Previous Patch Causes PJSIP Timer Issues
- Reported by: Ross Beer
- * [33ed2e1bb8] Joshua Colp -- pjproject-bundled: Add upstream timer
- fixes
- * [d4cd2a9706] Sean Bright -- pjproject: Add timer patch from pjproject
- r5934
-
- ----------------------------------------------------------------------
-
- Commits Not Associated with an Issue
-
- [Back to Top]
-
- This is a list of all changes that went into this release that did not
- reference a JIRA issue.
-
- +------------------------------------------------------------------------+
- | Revision | Author | Summary |
- |------------+-------------------+---------------------------------------|
- | | | Revert "res_rtp_asterisk: Free |
- | fe3dc091b5 | Joshua Colp | payload when error on insertion to |
- | | | data buffer" |
- |------------+-------------------+---------------------------------------|
- | a75317ce24 | Joshua Colp | Revert "res_rtp_asterisk: Resolve |
- | | | loop when receive buffer is flushed" |
- |------------+-------------------+---------------------------------------|
- | 81d34554ab | Asterisk | Update for certified/16.8-cert1-rc5 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | 25e2274e48 | Jaco Kroon | main/backtrace: binutils-2.34 fix. |
- |------------+-------------------+---------------------------------------|
- | 2af88e7ca2 | Jaco Kroon | Update main/backtrace.c to deal with |
- | | | changes in binutils 2.34. |
- |------------+-------------------+---------------------------------------|
- | 3d317239d5 | Asterisk | Update for certified/16.8-cert1-rc4 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | 096db19662 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | certified/16.8-cert1-rc4 |
- |------------+-------------------+---------------------------------------|
- | c9cd681261 | Joshua C. Colp | CHANGES: Change md file extension to |
- | | | txt. |
- |------------+-------------------+---------------------------------------|
- | 904f6b7c2f | Kevin Harwell | ast_coredumper: add Asterisk |
- | | | information dump |
- |------------+-------------------+---------------------------------------|
- | 7ecbcdf22f | George Joseph | CI: Create generic jenkinsfile |
- |------------+-------------------+---------------------------------------|
- | d1bb76a27d | Asterisk | Update for certified/16.8-cert1-rc3 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | b922e387d1 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | certified/16.8-cert1 |
- |------------+-------------------+---------------------------------------|
- | 7b8157645f | Asterisk | Update for certified/16.8-cert1-rc2 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | 476bbcf3a3 | Asterisk | Update for certified/16.8-cert1-rc1 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | b3a0387fc0 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | certified/16.8-cert1 |
- |------------+-------------------+---------------------------------------|
- | 7416703f04 | George Joseph | doc: Fix CHANGES entries to have .txt |
- | | | suffix and update READMEs |
- |------------+-------------------+---------------------------------------|
- | 443230f5d5 | George Joseph | Asterisk Certified 16.8 Preparation |
- |------------+-------------------+---------------------------------------|
- | 40e331ff90 | Joshua C. Colp | res_rtp_asterisk: Don't produce |
- | | | transport-cc if no packets. |
- |------------+-------------------+---------------------------------------|
- | 8b0922a101 | George Joseph | message.c: Add option to suppress the |
- | | | Message channel AMI and ARI events |
- |------------+-------------------+---------------------------------------|
- | d436f56858 | Asterisk | Update for 16.8.0 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | 126beb3e6c | Joshua Colp | REVERT: Add option to suppress the |
- | | | Message channel AMI and ARI events |
- |------------+-------------------+---------------------------------------|
- | bfe9e1b2e7 | George Joseph | message.c: Add option to suppress the |
- | | | Message channel AMI and ARI events |
- |------------+-------------------+---------------------------------------|
- | c92e2bb09f | Asterisk | Update for 16.8.0-rc2 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | b7b813eb34 | Asterisk | Update for 16.8.0-rc1 |
- | | Development Team | |
- |------------+-------------------+---------------------------------------|
- | eb1ec0498d | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.8.0 |
- |------------+-------------------+---------------------------------------|
- | a7aaca9eaa | Sean Bright | func_odbc.conf.sample: Add example |
- | | | lookup |
- |------------+-------------------+---------------------------------------|
- | f49517efb9 | Rodrigo RamÃrez | queue_log: Add alembic script for |
- | | Norambuena | generate db table for queue_log |
- |------------+-------------------+---------------------------------------|
- | 13fa33588f | Sean Bright | app_voicemail, say: Fix various |
- | | | leading whitespace problems |
- |------------+-------------------+---------------------------------------|
- | | | netsock2: |
- | b92b0469ff | Jaco Kroon | ast_addressfamily_to_sockaddrsize and |
- | | | ast_sockaddr_from_sockaddr. |
- |------------+-------------------+---------------------------------------|
- | de078debab | Kevin Harwell | app_agent_pool: Update XML docs for |
- | | | AgentLogin |
- |------------+-------------------+---------------------------------------|
- | 11753d94d8 | Richard Mudgett | features.c: Make Bridge application |
- | | | tolerate unspecified channel. |
- |------------+-------------------+---------------------------------------|
- | 00e745066c | Richard Mudgett | app_chanspy.c: Reduce log message |
- | | | level from notice to verbose. |
- |------------+-------------------+---------------------------------------|
- | 198f4cbdbf | Richard Mudgett | app_softhangup.c: Reduce unnecessary |
- | | | warning to verbose message. |
- |------------+-------------------+---------------------------------------|
- | efa13eb0a0 | Sean Bright | db: Initialize condition primitive |
- | | | before use |
- |------------+-------------------+---------------------------------------|
- | 9240fcd8bb | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.7.0 |
- |------------+-------------------+---------------------------------------|
- | 77941efad9 | Jaco Kroon | ACL: ast_apply_acl_nolog - identical |
- | | | to ast_apply_acl but without logging. |
- |------------+-------------------+---------------------------------------|
- | 9c29c3fb3e | Joshua Colp | Revert "PJSIP_CONTACT: add missing |
- | | | argument documentation" |
- |------------+-------------------+---------------------------------------|
- | | | res_pjsip_registrar.c: Prevent |
- | 5c20cc4c3a | Sean Bright | possible buffer overflow with domain |
- | | | aliases |
- |------------+-------------------+---------------------------------------|
- | 43d4c0e3c9 | Thomas Arimont | channel.c: Resolve issue with |
- | | | receiving SIP INFO packets for DTMF |
- |------------+-------------------+---------------------------------------|
- | 80199cd67f | George Joseph | CI: Turn off shallow cloning |
- | | | altogether |
- |------------+-------------------+---------------------------------------|
- | bd3cb1b300 | Sean Bright | media_cache.c: Various CLI |
- | | | improvements |
- |------------+-------------------+---------------------------------------|
- | 9eb86a8110 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.6.2 |
- |------------+-------------------+---------------------------------------|
- | d075d8913b | George Joseph | CI: Fix missing script block in |
- | | | jenkinsfiles |
- |------------+-------------------+---------------------------------------|
- | ce8a23fdf9 | George Joseph | CI: Fix missing script block in |
- | | | jenkinsfiles |
- |------------+-------------------+---------------------------------------|
- | 919bc0c7be | George Joseph | CI: Increase clone depth and do |
- | | | better cleanup |
- |------------+-------------------+---------------------------------------|
- | 61a3e7e79b | Sean Bright | res_pjsip_registrar: Fix uninitlized |
- | | | variable warning |
- |------------+-------------------+---------------------------------------|
- | 30c0af7257 | Kevin Harwell | various files - fix some alerts |
- | | | raised by lgtm code analysis |
- |------------+-------------------+---------------------------------------|
- | 6f0a69c51a | Kevin Harwell | res_pjsip_session: initialize |
- | | | pending's topology to endpoint's |
- |------------+-------------------+---------------------------------------|
- | 6be18dfb72 | Alexei Gradinari | serializer: set high/low alert levels |
- | | | on whole pool |
- |------------+-------------------+---------------------------------------|
- | bfd0e05e59 | George Joseph | ExternalMedia: Change return object |
- | | | from ExternalMedia to Channel |
- |------------+-------------------+---------------------------------------|
- | ef2eb064b2 | Joshua Colp | res_rtp_asterisk: Remove a log |
- | | | message that slipped in. |
- |------------+-------------------+---------------------------------------|
- | ed12715cbe | Joshua Colp | test_res_rtp: Enable FIR and REMB |
- | | | nominal tests. |
- |------------+-------------------+---------------------------------------|
- | 867c8b2879 | Chris Savinovich | test_taskprocessor.c: Fix test |
- | | | failure on Ubuntu |
- |------------+-------------------+---------------------------------------|
- | afc10c25ac | Kevin Harwell | serializer: move/add asterisk |
- | | | serializer pool functionality |
- |------------+-------------------+---------------------------------------|
- | 931ef77e21 | Kevin Harwell | res_pjsip/res_pjsip_mwi: use |
- | | | centralized serializer pools |
- |------------+-------------------+---------------------------------------|
- | 04f7d136d8 | Alexei Gradinari | res_pjsip_pubsub: add endpoint to |
- | | | some warning |
- |------------+-------------------+---------------------------------------|
- | d146ec7e83 | Asterisk Team | Revert "Update CHANGES and |
- | | | UPGRADE.txt for 16.6.0-rc2" |
- |------------+-------------------+---------------------------------------|
- | 85c7326d08 | Jonathan Rose | basic-pbx: Bring forward queue |
- | | | configuration from 13 |
- |------------+-------------------+---------------------------------------|
- | 827dd754b2 | Ben Ford | taskprocessor.c: Added "like" support |
- | | | to 'core show taskprocessors' |
- |------------+-------------------+---------------------------------------|
- | a0844a9b9b | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.6.0-rc2 |
- |------------+-------------------+---------------------------------------|
- | 9ff11c2f00 | Kevin Harwell | res_pjsip_pubsub: change warning to |
- | | | debug |
- |------------+-------------------+---------------------------------------|
- | cd51f5b876 | Corey Farrell | core: Fix ABI mismatch of |
- | | | ao2_global_obj. |
- |------------+-------------------+---------------------------------------|
- | 5ea667e03a | Ben Ford | taskprocessor.c: Add CLI commands to |
- | | | reset taskprocessor stats. |
- |------------+-------------------+---------------------------------------|
- | fec6e1bd87 | Corey Farrell | core: Add AO2_ALLOC_OPT_NO_REF_DEBUG |
- | | | option. |
- |------------+-------------------+---------------------------------------|
- | 9c5a8066a6 | George Joseph | astmm.c: Display backtrace with |
- | | | memory show allocations |
- |------------+-------------------+---------------------------------------|
- | 5b8c1ed0d3 | Corey Farrell | stasis: refcounter.py can incorrectly |
- | | | report skewed objects. |
- |------------+-------------------+---------------------------------------|
- | 76d4a42ae1 | Corey Farrell | res_pjsip_mwi: Remove inappropriate |
- | | | topic unreference. |
- |------------+-------------------+---------------------------------------|
- | 256db889f2 | Corey Farrell | app_voicemail: Fix module unload |
- | | | leak. |
- |------------+-------------------+---------------------------------------|
- | 9f304170f6 | Sean Bright | res_musiconhold: Add new 'playlist' |
- | | | mode |
- |------------+-------------------+---------------------------------------|
- | a95cef7140 | Ben Ford | res_rtp_asterisk.c: Send RTCP as |
- | | | compound packets. |
- |------------+-------------------+---------------------------------------|
- | 688908fe7a | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.6.0 |
- |------------+-------------------+---------------------------------------|
- | 922d3e02df | Ben Ford | res_rtp: Add unit tests for RTCP |
- | | | stats. |
- |------------+-------------------+---------------------------------------|
- | d566314e38 | George Joseph | ARI: External Media |
- |------------+-------------------+---------------------------------------|
- | | | test_utils.c: Skip test |
- | a321225fa4 | Chris-Savinovich | adsi_loaded_test if module not |
- | | | loaded. |
- |------------+-------------------+---------------------------------------|
- | 78d00c277c | Igor Goncharovsky | chan_unistim: Fix code, causing all |
- | | | incoming DTMF sent back to asterisk |
- |------------+-------------------+---------------------------------------|
- | 821b7561f8 | Igor Goncharovsky | chan_unistim: Fix RTP port byte order |
- | | | for big-endian arch |
- |------------+-------------------+---------------------------------------|
- | aaaa1695ca | Alexei Gradinari | Fix misname 'res_external_mwi' to |
- | | | 'res_mwi_external' in comments. |
- |------------+-------------------+---------------------------------------|
- | c00a010fe8 | George Joseph | chan_rtp: Accept hostname as well as |
- | | | ip address as destination |
- |------------+-------------------+---------------------------------------|
- | 6407ccd2d9 | George Joseph | dns_core: Create new API |
- | | | ast_dns_resolve_ipv6_and_ipv4 |
- |------------+-------------------+---------------------------------------|
- | f82d0b74fd | George Joseph | res_ari.c: Prefer exact handler match |
- | | | over wildcard |
- |------------+-------------------+---------------------------------------|
- | 51fd43206b | Sean Bright | audiohook.c: Substitute silence for |
- | | | unavailable audio frames |
- |------------+-------------------+---------------------------------------|
- | 92066b8746 | George Joseph | CI: Escape backslashes in |
- | | | printenv/sort/tr |
- |------------+-------------------+---------------------------------------|
- | db9684ad1e | George Joseph | CI: Add "throttle" label and |
- | | | "skip_gate" capability |
- |------------+-------------------+---------------------------------------|
- | 2641081caa | George Joseph | CI: Make node labels job-specific |
- |------------+-------------------+---------------------------------------|
- | 9718376902 | Sean Bright | res_musiconhold: Use a vector instead |
- | | | of custom array allocation |
- |------------+-------------------+---------------------------------------|
- | 0ebfc4a19d | Sean Bright | manager: Send fewer packets |
- |------------+-------------------+---------------------------------------|
- | d6af1acb8c | Sean Bright | res_musiconhold: Use |
- | | | ast_pipe_nonblock() wrapper |
- |------------+-------------------+---------------------------------------|
- | 05cf9c9912 | George Joseph | loader.c: Fix possible SEGV when a |
- | | | module fails to register |
- |------------+-------------------+---------------------------------------|
- | 06780d2bc4 | George Joseph | CI: Don't enable non-core modules in |
- | | | Certified branches |
- |------------+-------------------+---------------------------------------|
- | 3814faf848 | Leonid Fainshtein | openr2(6/6): Set hangup cause |
- |------------+-------------------+---------------------------------------|
- | 06515707df | Tzafrir Cohen | openr2(5/6): added cli command -- |
- | | | mfcr2 destroy link <index> |
- |------------+-------------------+---------------------------------------|
- | 93a093f6c4 | Tzafrir Cohen | openr2(4/6): added new cli command -- |
- | | | mfcr2 show links |
- |------------+-------------------+---------------------------------------|
- | a45cfefb77 | Tzafrir Cohen | openr2(3/6): Convert r2links to |
- | | | standard Asterisk AST_LIST* |
- |------------+-------------------+---------------------------------------|
- | | | openr2(2/6): Stop polling channels |
- | ec6e88592a | Tzafrir Cohen | when DAHDI returns -ENODEV (e.g: |
- | | | plug-out) |
- |------------+-------------------+---------------------------------------|
- | 7b6df814a6 | Tzafrir Cohen | openr2(1/6): bugfix in configuration |
- | | | saving |
- |------------+-------------------+---------------------------------------|
- | 356f4256cc | George Joseph | CI: Add cleanWs to cleanup steps in |
- | | | jenkinsfiles |
- |------------+-------------------+---------------------------------------|
- | 9d69469203 | Rodrigo RamÃrez | README.md: Update year |
- | | Norambuena | |
- |------------+-------------------+---------------------------------------|
- | c86c0973ff | George Joseph | CI: Add install-headers to the |
- | | | install make targets |
- |------------+-------------------+---------------------------------------|
- | cb831a18d1 | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.5.0 |
- |------------+-------------------+---------------------------------------|
- | f62d9013c1 | George Joseph | Build: Add separate header |
- | | | install/uninstall targets |
- |------------+-------------------+---------------------------------------|
- | 88ea395c33 | Kevin Harwell | manager: Log AMI actions |
- |------------+-------------------+---------------------------------------|
- | 1756029237 | Joshua Colp | res_rtp_asterisk: Move where DTLS MTU |
- | | | variable is defined. |
- |------------+-------------------+---------------------------------------|
- | 31d755e805 | George Joseph | sig_pri: Address gcc9 issues |
- |------------+-------------------+---------------------------------------|
- | 01712bbdc9 | George Joseph | CI: New way to determnine libdir |
- |------------+-------------------+---------------------------------------|
- | 1ee2f01f62 | George Joseph | chan_dahdi: Address gcc9 issues |
- |------------+-------------------+---------------------------------------|
- | 8b77318a2c | Alexei Gradinari | translate.c do not log WARNING on |
- | | | empty audio frame |
- |------------+-------------------+---------------------------------------|
- | ccc92b6ecb | George Joseph | app_confbridge: Attended transfer |
- | | | event fixup |
- |------------+-------------------+---------------------------------------|
- | 694097ee68 | Sean Bright | pjproject: Update to 2.9 release |
- |------------+-------------------+---------------------------------------|
- | 86cd77ec0a | Alexei Gradinari | app_attended_transfer: new |
- | | | application AttendedTransfer |
- |------------+-------------------+---------------------------------------|
- | 6321b559b9 | Alexei Gradinari | res_fax: gateway sends T.38 request |
- | | | to both endpoints if V.21 detected |
- |------------+-------------------+---------------------------------------|
- | d2c07aceca | Asterisk | Update CHANGES and UPGRADE.txt for |
- | | Development Team | 16.4.0 |
- |------------+-------------------+---------------------------------------|
- | e77704f45c | Alexei Gradinari | res_fax: add channel name to CLI 'fax |
- | | | show session' |
- |------------+-------------------+---------------------------------------|
- | ec74fd56a7 | Ben Ford | build: Fix file format in |
- | | | CHANGES-staging. |
- |------------+-------------------+---------------------------------------|
- | db5bc0fabf | Alexei Gradinari | app_blind_transfer: new application |
- | | | BlindTransfer |
- |------------+-------------------+---------------------------------------|
- | 9516fb64c9 | Alexei Gradinari | app_readexten: new option 'p' to stop |
- | | | reading on '#' key |
- |------------+-------------------+---------------------------------------|
- | 79b15d0b30 | George Joseph | res_rtp_asterisk: Add ability to |
- | | | propose local address in ICE |
- |------------+-------------------+---------------------------------------|
- | 941dead08d | Ben Ford | pjsip_options.c: Allow immediate |
- | | | qualifies for new contacts. |
- |------------+-------------------+---------------------------------------|
- | edc3e0df1a | Kevin Harwell | conversions.c: Add conversions for |
- | | | largest max sized integer |
- |------------+-------------------+---------------------------------------|
- | e3a758975d | Kevin Harwell | mwi core: Move core MWI functionality |
- | | | into its own files |
- |------------+-------------------+---------------------------------------|
- | e281911667 | George Joseph | ARI: Bump non-breaking version number |
- | | | to 4.0.2 |
- |------------+-------------------+---------------------------------------|
- | 7487fc88d2 | George Joseph | res_remb_modifier: Propertly |
- | | | initialize bitrate to 0.0 |
- |------------+-------------------+---------------------------------------|
- | | | res_mwi_devstate: Specify |
- | 022e784b7a | Sean Bright | AST_MODFLAG_LOAD_ORDER to enable load |
- | | | priority |
- |------------+-------------------+---------------------------------------|
- | 18fe583d12 | George Joseph | CI: Move test group config files to |
- | | | Jenkins |
- |------------+-------------------+---------------------------------------|
- | 898765d919 | George Joseph | ARI: Run 'make ari-stubs' |
- |------------+-------------------+---------------------------------------|
- | acfbfef8ad | Alexei Gradinari | res_pjsip: Fix transport_states ref |
- | | | leak |
- |------------+-------------------+---------------------------------------|
- | 85bbb7a3e8 | Chris-Savinovich | config.c: Fix a crash in extconfig |
- | | | parsing |
- |------------+-------------------+---------------------------------------|
- | 1464a6b80f | George Joseph | CI: Add --no-dev-mode option to |
- | | | buildAsterisk.sh |
- |------------+-------------------+---------------------------------------|
- | ae1aeb930e | Matthew | res/res_rtp_asterisk: Enable rxjitter |
- | | Fredrickson | calculation for video |
- +------------------------------------------------------------------------+
-
- ----------------------------------------------------------------------
-
- 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.
-
- UPGRADE-1.2.txt | 218
- UPGRADE-1.4.txt | 497
- UPGRADE-1.6.txt | 277
- UPGRADE-1.8.txt | 343
- UPGRADE-10.txt | 92
- UPGRADE-11.txt | 280
- UPGRADE-12.txt | 478
- UPGRADE-13.txt | 399
- UPGRADE-14.txt | 115
- UPGRADE-15.txt | 63
- asterisk-certified-16.3-cert1-summary.html | 7965 ----
- asterisk-certified-16.3-cert1-summary.txt |19102 ----------
- b/.gitreview | 2
- b/.version | 2
- b/CHANGES | 270
- b/ChangeLog | 4334 ++
- b/Makefile | 30
- b/Makefile.rules | 9
- b/README-SERIOUSLY.bestpractices.md | 4
- b/README.md | 2
- b/UPGRADE.txt | 2611 +
- b/addons/cdr_mysql.c | 18
- b/addons/chan_ooh323.c | 19
- b/addons/chan_ooh323.h | 1
- b/addons/ooh323c/src/ooh323ep.c | 6
- b/addons/ooh323c/src/ooh323ep.h | 2
- b/apps/app_adsiprog.c | 2
- b/apps/app_agent_pool.c | 8
- b/apps/app_amd.c | 32
- b/apps/app_attended_transfer.c | 144
- b/apps/app_blind_transfer.c | 138
- b/apps/app_cdr.c | 8
- b/apps/app_chanisavail.c | 137
- b/apps/app_chanspy.c | 3
- b/apps/app_confbridge.c | 163
- b/apps/app_dial.c | 138
- b/apps/app_dictate.c | 4
- b/apps/app_followme.c | 12
- b/apps/app_minivm.c | 4
- b/apps/app_mixmonitor.c | 13
- b/apps/app_page.c | 30
- b/apps/app_playback.c | 9
- b/apps/app_queue.c | 20
- b/apps/app_readexten.c | 14
- b/apps/app_record.c | 3
- b/apps/app_senddtmf.c | 13
- b/apps/app_softhangup.c | 2
- b/apps/app_voicemail.c | 580
- b/apps/confbridge/conf_config_parser.c | 35
- b/apps/confbridge/confbridge_manager.c | 27
- b/apps/confbridge/include/confbridge.h | 14
- b/asterisk-certified-16.8-cert1-rc5-summary.html | 16
- b/asterisk-certified-16.8-cert1-rc5-summary.txt | 101
- b/bridges/bridge_native_rtp.c | 173
- b/bridges/bridge_simple.c | 198
- b/bridges/bridge_softmix.c | 377
- b/bridges/bridge_softmix/include/bridge_softmix_internal.h | 2
- b/cdr/cdr_pgsql.c | 26
- b/cel/cel_pgsql.c | 2
- b/channels/chan_console.c | 4
- b/channels/chan_dahdi.c | 480
- b/channels/chan_dahdi.h | 19
- b/channels/chan_iax2.c | 29
- b/channels/chan_mgcp.c | 1
- b/channels/chan_motif.c | 9
- b/channels/chan_pjsip.c | 200
- b/channels/chan_rtp.c | 19
- b/channels/chan_sip.c | 43
- b/channels/chan_skinny.c | 1
- b/channels/chan_unistim.c | 175
- b/channels/pjsip/cli_commands.c | 13
- b/channels/pjsip/dialplan_functions.c | 65
- b/channels/pjsip/include/dialplan_functions.h | 25
- b/channels/sig_analog.c | 7
- b/channels/sig_pri.c | 33
- 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 | 5
- b/configs/samples/confbridge.conf.sample | 6
- b/configs/samples/dundi.conf.sample | 6
- b/configs/samples/extconfig.conf.sample | 1
- b/configs/samples/func_odbc.conf.sample | 8
- b/configs/samples/http.conf.sample | 10
- b/configs/samples/musiconhold.conf.sample | 23
- b/configs/samples/pjsip.conf.sample | 6
- b/configs/samples/rtp.conf.sample | 4
- b/configure | 192
- b/configure.ac | 36
- b/contrib/ast-db-manage/README.md | 1
- b/contrib/ast-db-manage/config/versions/3a094a18e75b_pjsip_add_norefersub.py | 39
- 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 | 21
- b/contrib/realtime/postgresql/postgresql_config.sql | 27
- b/contrib/scripts/ast_coredumper | 417
- b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
- b/contrib/valgrind.supp | 14
- b/doc/CHANGES-staging/README.md | 28
- b/doc/UPGRADE-staging/README.md | 31
- b/doc/appdocsxml.dtd | 2
- b/formats/format_g726.c | 16
- b/formats/msgsm.h | 4
- b/funcs/func_curl.c | 48
- b/funcs/func_env.c | 5
- b/funcs/func_jitterbuffer.c | 19
- b/funcs/func_odbc.c | 22
- b/funcs/func_pjsip_contact.c | 6
- b/funcs/func_talkdetect.c | 2
- b/include/asterisk/abstract_jb.h | 4
- b/include/asterisk/acl.h | 37
- b/include/asterisk/app.h | 195
- b/include/asterisk/ari.h | 2
- b/include/asterisk/astobj2.h | 5
- b/include/asterisk/audiohook.h | 2
- b/include/asterisk/autoconfig.h.in | 12
- b/include/asterisk/bridge.h | 9
- b/include/asterisk/calendar.h | 4
- b/include/asterisk/cel.h | 22
- b/include/asterisk/channel.h | 44
- 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 | 20
- b/include/asterisk/dns_core.h | 22
- b/include/asterisk/dns_internal.h | 5
- b/include/asterisk/format_cache.h | 5
- b/include/asterisk/http_websocket.h | 14
- b/include/asterisk/iostream.h | 14
- b/include/asterisk/json.h | 16
- b/include/asterisk/max_forwards.h | 1
- b/include/asterisk/mixmonitor.h | 5
- b/include/asterisk/mwi.h | 242
- b/include/asterisk/netsock2.h | 42
- b/include/asterisk/options.h | 3
- b/include/asterisk/parking.h | 5
- b/include/asterisk/res_fax.h | 3
- b/include/asterisk/res_pjsip.h | 12
- b/include/asterisk/res_pjsip_presence_xml.h | 5
- b/include/asterisk/res_pjsip_pubsub.h | 23
- b/include/asterisk/res_pjsip_session.h | 2
- b/include/asterisk/rtp_engine.h | 129
- b/include/asterisk/sched.h | 16
- b/include/asterisk/serializer.h | 85
- b/include/asterisk/slin.h | 5
- b/include/asterisk/stasis.h | 46
- b/include/asterisk/stasis_bridges.h | 23
- b/include/asterisk/taskprocessor.h | 9
- b/include/asterisk/utils.h | 9
- b/main/abstract_jb.c | 178
- b/main/acl.c | 74
- b/main/aoc.c | 8
- b/main/app.c | 338
- b/main/asterisk.c | 18
- b/main/astmm.c | 23
- b/main/astobj2.c | 88
- b/main/astobj2_container.c | 24
- b/main/astobj2_global.c | 97
- b/main/astobj2_hash.c | 21
- b/main/astobj2_rbtree.c | 13
- b/main/audiohook.c | 11
- b/main/backtrace.c | 9
- b/main/bridge.c | 1
- b/main/ccss.c | 4
- b/main/cdr.c | 65
- b/main/cel.c | 82
- b/main/channel.c | 80
- b/main/codec_builtin.c | 8
- b/main/config.c | 33
- b/main/conversions.c | 39
- b/main/core_local.c | 6
- b/main/db.c | 3
- b/main/dns_core.c | 72
- b/main/event.c | 17
- b/main/features.c | 28
- b/main/file.c | 44
- b/main/format_cache.c | 8
- b/main/http.c | 56
- b/main/indications.c | 10
- b/main/iostream.c | 14
- b/main/json.c | 17
- b/main/loader.c | 149
- b/main/manager.c | 120
- b/main/manager_mwi.c | 1
- b/main/media_cache.c | 51
- b/main/message.c | 6
- b/main/mwi.c | 369
- b/main/options.c | 2
- b/main/pbx.c | 71
- b/main/pbx_variables.c | 23
- b/main/rtp_engine.c | 253
- b/main/say.c | 956
- b/main/sched.c | 30
- b/main/serializer.c | 189
- b/main/stasis.c | 479
- b/main/stasis_bridges.c | 36
- b/main/stasis_cache.c | 10
- b/main/stasis_channels.c | 9
- b/main/stream.c | 22
- b/main/taskprocessor.c | 219
- b/main/tcptls.c | 24
- b/main/translate.c | 13
- b/main/udptl.c | 18
- b/menuselect/Makefile | 8
- b/menuselect/autoconfig.h.in | 3
- b/menuselect/configure | 348
- b/menuselect/configure.ac | 12
- b/menuselect/example_menuselect-tree | 4
- b/menuselect/makeopts.in | 3
- b/menuselect/test/menuselect-tree | 4
- b/pbx/pbx_dundi.c | 135
- b/res/ael/pval.c | 30
- b/res/ari/ari_model_validators.c | 412
- b/res/ari/ari_model_validators.h | 55
- b/res/ari/config.c | 10
- b/res/ari/resource_channels.c | 216
- b/res/ari/resource_channels.h | 55
- b/res/ari/resource_events.c | 10
- b/res/parking/parking_bridge_features.c | 2
- b/res/parking/res_parking.h | 5
- b/res/res_agi.c | 20
- b/res/res_ari.c | 23
- b/res/res_ari_channels.c | 206
- b/res/res_ari_events.c | 2
- b/res/res_calendar_ews.c | 1
- b/res/res_calendar_exchange.c | 1
- b/res/res_calendar_icalendar.c | 1
- b/res/res_config_curl.c | 5
- b/res/res_config_pgsql.c | 2
- b/res/res_config_sqlite3.c | 6
- b/res/res_corosync.c | 2
- b/res/res_fax.c | 81
- b/res/res_http_websocket.c | 11
- b/res/res_musiconhold.c | 294
- b/res/res_mwi_devstate.c | 4
- b/res/res_mwi_external.c | 1
- b/res/res_phoneprov.c | 6
- b/res/res_pjsip.c | 94
- b/res/res_pjsip/config_global.c | 21
- b/res/res_pjsip/config_system.c | 2
- b/res/res_pjsip/config_transport.c | 17
- b/res/res_pjsip/location.c | 6
- b/res/res_pjsip/pjsip_configuration.c | 4
- b/res/res_pjsip/pjsip_message_filter.c | 53
- b/res/res_pjsip/pjsip_options.c | 47
- b/res/res_pjsip/pjsip_resolver.c | 4
- b/res/res_pjsip_config_wizard.c | 7
- b/res/res_pjsip_dialog_info_body_generator.c | 80
- b/res/res_pjsip_endpoint_identifier_ip.c | 102
- b/res/res_pjsip_mwi.c | 590
- b/res/res_pjsip_nat.c | 84
- b/res/res_pjsip_notify.c | 22
- b/res/res_pjsip_outbound_registration.c | 54
- b/res/res_pjsip_publish_asterisk.c | 2
- b/res/res_pjsip_pubsub.c | 107
- b/res/res_pjsip_refer.c | 5
- b/res/res_pjsip_registrar.c | 59
- b/res/res_pjsip_sdp_rtp.c | 29
- b/res/res_pjsip_session.c | 178
- b/res/res_pjsip_t38.c | 40
- b/res/res_pjsip_transport_websocket.c | 4
- b/res/res_realtime.c | 56
- b/res/res_remb_modifier.c | 67
- b/res/res_resolver_unbound.c | 6
- b/res/res_rtp_asterisk.c | 1214
- b/res/res_smdi.c | 1
- b/res/res_srtp.c | 33
- b/res/res_stasis.c | 45
- b/res/res_statsd.c | 35
- b/res/res_xmpp.c | 1
- b/res/stasis/app.c | 15
- b/res/stasis/command.c | 2
- b/res/stasis/control.c | 19
- b/rest-api-templates/res_ari_resource.c.mustache | 2
- b/rest-api/api-docs/channels.json | 329
- b/rest-api/resources.json | 2
- b/tests/CI/buildAsterisk.sh | 24
- b/tests/CI/findLibdir.sh | 14
- b/tests/CI/gateTestGroups.json | 7
- b/tests/CI/gates.jenkinsfile | 32
- b/tests/CI/installAsterisk.sh | 2
- b/tests/CI/periodic-dailyTestGroups.json | 6
- b/tests/CI/periodics-daily.jenkinsfile | 19
- b/tests/CI/ref_debug.jenkinsfile | 10
- b/tests/CI/unittests.jenkinsfile | 10
- b/tests/CI/universal-asterisk-nongerrit.jenkinsfile | 452
- b/tests/test_conversions.c | 48
- b/tests/test_data_buffer.c | 2
- b/tests/test_json.c | 18
- b/tests/test_res_rtp.c | 516
- b/tests/test_stasis_channels.c | 4
- b/tests/test_taskprocessor.c | 78
- b/third-party/jansson/Makefile | 7
- b/third-party/pjproject/Makefile | 18
- b/third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157
- b/third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch | 53
- b/third-party/pjproject/patches/0030-ssl-regression-fix.patch | 105
- b/third-party/pjproject/patches/0031-transport-regression-fix.patch | 187
- doc/CHANGES-staging/app_confbridge_maximum_sample_rate.txt | 5
- doc/CHANGES-staging/rtp_ice_include_local_address.txt | 5
- doc/UPGRADE-staging/AMI-Originate.txt | 5
- third-party/pjproject/patches/0010-outgoing_connected_line_method_update.patch | 33
- third-party/pjproject/patches/0020-Fixed-2172-Avoid-double-reference-counter-decrements.patch | 42
- third-party/pjproject/patches/0031-Re-2191-transport-timer-cleanup.patch | 372
- third-party/pjproject/patches/0032-Re-2191-Fixed-crash-in-SIP-transport-destroy-due-to-.patch | 131
- 316 files changed, 21717 insertions(+), 34013 deletions(-)
projects / thirdparty / asterisk.git / commitdiff
