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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/13.8-cert1-rc1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-certified/13.8-cert1-rc1</h3><h3 align="center">Date: 2016-04-06</h3><h3 align="center">&lt;asteriskteam@digium.com&gt;</h3><hr><h2 align="center">Table of Contents</h2><ol>
<li><a href="#summary">Summary</a></li>
<li><a href="#contributors">Contributors</a></li>
<li><a href="#closed_issues">Closed Issues</a></li>
<li><a href="#open_issues">Open Issues</a></li>
<li><a href="#commits">Other Changes</a></li>
<li><a href="#diffstat">Diffstat</a></li>
</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-certified/13.7.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
<tr valign="top"><td width="33%">215 Richard Mudgett <rmudgett@digium.com><br/>108 Matt Jordan <mjordan@digium.com><br/>103 Joshua Colp <jcolp@digium.com><br/>100 gtjoseph <george.joseph@fairview5.com><br/>87 Mark Michelson <mmichelson@digium.com><br/>67 Corey Farrell <git@cfware.com><br/>51 Kevin Harwell <kharwell@digium.com><br/>27 Scott Griepentrog <scott@griepentrog.com><br/>19 Alexander Traud <pabstraud@compuserve.com><br/>18 Diederik de Groot <dkgroot@talon.nl> (License 6600)<br/>13 Jonathan Rose <jrose@digium.com><br/>13 Walter Doekes <walter+asterisk@wjd.nu><br/>12 Rodrigo Ramírez Norambuena <a@rodrigoramirez.com><br/>11 Kinsey Moore <kmoore@digium.com><br/>10 Diederik de Groot <ddegroot@talon.nl><br/>10 David M. Lee <dlee@respoke.io><br/>10 Ivan Poddubny <ivan.poddubny@gmail.com><br/>7 Rusty Newton <rnewton@digium.com><br/>7 Benjamin Ford <bford@digium.com><br/>6 Ashley Sanders <asanders@digium.com><br/>5 Mark Michelson <mmichelson@lunkwill><br/>5 Sean Bright <sean.bright@gmail.com><br/>4 Dade Brandon <dade@xencall.com><br/>4 snuffy <snuffy22@gmail.com><br/>3 Badalyan Vyacheslav <slavon.net@gmail.com><br/>3 Martin Tomec <tomec.martin@gmail.com><br/>3 ibercom <ibercom123@gmail.com><br/>3 Graham Barnett (License 6685)<br/>3 Daniel Journo <dan@keshercommunications.com><br/>3 Scott Emidy <jemidy@digium.com><br/>2 Yousf Ateya <y.ateya@starkbits.com><br/>2 Steve Davies <steve@one47.co.uk><br/>2 Alexander Anikin <may213@yandex.ru><br/>2 Karsten Wemheuer <kwe-digium@iptam.com><br/>2 yaron nahum (License 6676)<br/>2 Tyler Cambron <tcambron@digium.com><br/>2 Stefan Engström <stefanen@kth.se><br/>2 Niklas Larsson <niklas@tese.se><br/>1 LEI FU (License 6640)<br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Michael L. Young (license 5026)<br/>1 Aaron An <anjb@ti-net.com.cn><br/>1 Damian Ivereigh <damo@launtel.net.au><br/>1 Andreas Steinmetz (license 6523)<br/>1 Alexei Gradinari <alex2grad@gmail.com><br/>1 Christof Lauber <christof.lauber@annax.ch><br/>1 demon-ru <serov.d.p@gmail.com><br/>1 Florian Sauerteig <ffs@ccn.net><br/>1 Olle Johansson (License 5267)<br/>1 Alexei Gradinari License #5691<br/>1 Makoto Dei (License 5027)<br/>1 Eugene Voityuk <eugene@thirdlane.com><br/>1 Filip Jenicek <phill@janevim.cz><br/>1 Valentin Vidić (License 6697)<br/>1 Carlos Oliva <carlos.oliva@invoxcontact.com><br/>1 Olle Johansson <oej@edvina.net> (License 5267)<br/>1 Kristian Hogh (License 6639)<br/>1 Ben Klang (License 5876)<br/>1 Alexandre Fournier <alexandre.fournier@kiplink.fr><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Di-Shi Sun (License 5076)<br/>1 Ed Hynan (Licnese 6680)<br/>1 D Tucny <d@tucny.com><br/>1 Javier Acosta (License 6690)<br/>1 Etienne Lessard (license #6394)<br/>1 HZMI8gkCvPpom0tM (License 6658)<br/>1 Y Ateya (License 6693)<br/>1 sungtae kim <pchero21@gmail.com><br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Gareth Palmer (License 5169)<br/>1 Corey Edwards <tensai@zmonkey.org><br/>1 Maciej Szmigiero <mail@maciej.szmigiero.name> (license 6085)<br/>1 Ben Merrills (License 6678)<br/>1 Justin T. Gibbs <gibbs@scsiguy.org> (License 6692)<br/>1 server-pandora <server-pandora@xencall.com><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Jaco Kroon (License 5671)<br/>1 Matt Hoskins (license 6688)<br/>1 Andrew Nagy <andrew.nagy@the159.com><br/>1 cloos <cloos@jhcloos.com> (License #5956)<br/>1 Matthias Urlichs (license 5508)<br/>1 Stefan Engström (License 6691)<br/>1 abelbeck <lonnie@abelbeck.com> (License 5903)<br/>1 Simon Arlott (License 5756)<br/>1 Richard Miller (License 5685)<br/>1 Patric Marschall <patric.marschall@1und1.de><br/>1 Mark Duncan <mark@syon.co.jp><br/>1 Alec Davis <sivad.a@paradise.net.nz><br/>1 Debian Amtelco <dan@amtelco.com><br/>1 Juergen Spies (License 6698)<br/>1 Jeremiah Gowdy (License 6358)<br/>1 Sergio Medina Toledo <lumasepa@gmail.com><br/>1 Michael Cargile <mikec@vicidial.com><br/>1 Leif Madsen <leif@leifmadsen.com><br/></td><td width="33%">47 gtjoseph <george.joseph@fairview5.com><br/>4 Rusty Newton<br/>3 Badalyan Vyacheslav<br/>3 Matt Jordan <mjordan@digium.com><br/>2 snuffy <snuffy22@gmail.com><br/>2 Stefan Engström<br/>1 Sebastian Kemper<br/>1 JoshE<br/>1 Dmitriy Serov<br/>1 Michael L. Young<br/>1 starting asterisk -c until the colors stopped<br/>1 Graham Barnett<br/>1 Jeremiah Gowdy<br/>1 Aaron An<br/>1 XenCALL<br/>1 Kristian Høgh<br/>1 Damian Ivereigh<br/>1 Ben Klang<br/>1 Jacek Konieczny<br/>1 Alexander Traud<br/>1 Ivan Poddubny<br/>1 Corey Edwards <tensai@zmonkey.org><br/>1 Carl Fortin<br/>1 Juergen Spies<br/>1 Elazar Broad<br/>1 Alexandre Fournier<br/>1 Dan Cropp<br/>1 Matt Hoskins<br/>1 Di-Shi Sun<br/>1 Ed Hynan<br/></td><td width="33%">73 Matt Jordan <mjordan@digium.com><br/>49 Corey Farrell <git@cfware.com><br/>38 Joshua Colp <jcolp@digium.com><br/>34 Kevin Harwell <kharwell@digium.com><br/>33 Richard Mudgett <rmudgett@digium.com><br/>31 Mark Michelson<br/>28 Diederik de Groot <dkgroot@talon.nl><br/>28 Mark Michelson <mmichelson@digium.com><br/>19 George Joseph <george.joseph@fairview5.com><br/>19 Alexander Traud <pabstraud@compuserve.com><br/>15 Scott Griepentrog <sgriepentrog@digium.com><br/>15 Rusty Newton <rnewton@digium.com><br/>15 Richard Mudgett<br/>13 Arnd Schmitter <aschmitter@megasat.de><br/>13 gtjoseph<br/>12 Badalian Vyacheslav <slavon.net@gmail.com><br/>11 Kevin Harwell<br/>11 John Bigelow<br/>10 Walter Doekes <walter+asterisk@wjd.nu><br/>10 John Bigelow <jbigelow@digium.com><br/>10 Joshua Colp<br/>10 Dmitriy Serov <serov.d.p@gmail.com><br/>9 Rodrigo Ramirez Norambuena <a@rodrigoramirez.com><br/>9 Rusty Newton<br/>9 John Hardin<br/>8 Jonathan Rose <jrose@digium.com><br/>7 Stefan Engström <stefanen@kth.se><br/>6 Steve Pitts<br/>6 Andrew Nagy <andrew.nagy@the159.com><br/>5 Etienne Lessard <elessard@proformatique.com><br/>5 Michael Keuter <lists@mksolutions.info><br/>5 Jonathan Rose<br/>5 yaron nahum <nachum.yaron@gmail.com><br/>5 Chet Stevens <cwstevens@interact.ccsd.net><br/>5 Etienne Lessard<br/>5 Niklas Larsson <niklas@tese.se><br/>5 Scott Griepentrog<br/>5 snuffy <snuffy22@gmail.com><br/>5 Badalian Vyacheslav<br/>5 Andrew Nagy<br/>5 Ashley Sanders<br/>4 Ashley Sanders <asanders@digium.com><br/>4 Carl Fortin <cfortin2@cegepgarneau.ca><br/>4 Alexander Traud<br/>4 Ross Beer <ross.beer@voicehost.co.uk><br/>4 Chet Stevens<br/>4 Carl Fortin<br/>4 yaron nahum<br/>4 Dade Brandon <dade@xencall.com><br/>3 Steve Davies <steve@one47.co.uk><br/>3 Graham Barnett <graham@system-development.co.uk><br/>3 Zane Conkle <zconkle@cytracom.com><br/>3 Vitezslav Novy <a1@vnovy.net><br/>3 Niklas Larsson<br/>3 hristo <htrendev@gmail.com><br/>3 Zane Conkle<br/>3 Dmitriy Serov<br/>3 JoshE <josh@fluentstream.com><br/>3 Daniel Journo <dan@keshercommunications.com><br/>3 Y Ateya <y.ateya@starkbits.com><br/>2 tootai <admin@tootai.net><br/>2 warren smith <warren@serverplus.com><br/>2 Sean Bright <sean.bright@gmail.com><br/>2 Ivan Poddubny <ivan.poddubny@gmail.com><br/>2 Graham Barnett<br/>2 Rodrigo Ramirez Norambuena<br/>2 Alexandr Dranchuk <alex.dranchuk@gmail.com><br/>2 Marcelo Terres <mhterres@gmail.com><br/>2 Richard Kenner <kenner@gnat.com><br/>2 Makoto Dei<br/>2 Ray Crumrine <hraycrum-proftech@yahoo.com><br/>2 Makoto Dei <makotod@gmail.com><br/>2 Badalyan Vyacheslav<br/>2 Marcelo Terres<br/>2 Ross Beer<br/>2 JoshE<br/>2 John Zhong <john138@gmail.com><br/>2 Y Ateya<br/>2 Ray Crumrine<br/>2 Diederik de Groot<br/>2 David M. Lee <dlee@digium.com><br/>2 Denis Martinez<br/>2 David Brillert <david_brillert@scopserv.com><br/>2 ibercom <ibercom123@gmail.com><br/>2 Kinsey Moore <kmoore@digium.com><br/>2 Javier Riveros <goseeped@gmail.com><br/>2 cloos <cloos@jhcloos.com><br/>2 Daniel Journo<br/>2 Vadim <vadimd333@gmail.com><br/>2 Richard Kenner<br/>2 Vitezslav Novy<br/>2 Sean Pimental<br/>2 Karsten Wemheuer <kwe-digium@iptam.com><br/>1 Oleg Kozlov <olkeep@gmail.com><br/>1 Yura Kocyuba <yurakocyuba@yandex.ru><br/>1 Sebastian Kemper <sebastian_ml@gmx.net><br/>1 Damian Ivereigh <damo@launtel.net.au><br/>1 Ivan Poddubny<br/>1 Alexandr Gordeev<br/>1 ffs <ffs@ccn.net><br/>1 Jaco Kroon <jaco@uls.co.za><br/>1 Terry Wilson<br/>1 Rodrigo Ramírez Norambuena <decipher.hk@gmail.com><br/>1 Alejandro Mejia <amejia@gua.net><br/>1 Jeffrey Ollie<br/>1 Marcel Manz <marcel.manz@simmcomm.ch><br/>1 Terry Wilson <otherwiseguy@gmail.com><br/>1 Benjamin Keith Ford <bford@digium.com><br/>1 Alexandr Dranchuk<br/>1 Josh Kitchens<br/>1 Javier Acosta<br/>1 pj <silfex@centrum.cz><br/>1 Olivier Krief<br/>1 Dmitry Burilov <netaskd@gmail.com><br/>1 Aaron An<br/>1 Anthony Messina<br/>1 XenCALL<br/>1 Nir Simionovich (GreenfieldTech - Israel) <info-nospam@greenfieldtech.net><br/>1 Matt Hoskins <matt.hoski@gmail.com><br/>1 Aaron An <anjb@ti-net.com.cn><br/>1 Bryant Zimmerman <bryantz@zktech.com><br/>1 WRP <taylor@wrprojects.com><br/>1 Gareth Blades <gareth.blades@skymarket.co.uk><br/>1 Andreas Steinmetz<br/>1 Nic Colledge<br/>1 Nick Ruggles <nick.ruggles@gagenetworks.com><br/>1 ibercom <ibercom123@gmail.com><br/>1 Warren Selby<br/>1 Frank DiGennaro<br/>1 Mitch Claborn <mitch@claborn.net><br/>1 Jeremy Kister <asterisk.org@jeremykister.com><br/>1 Philippe Bolduc <philippe@florenceinc.com><br/>1 Alex A. Welzl <a.welzl@sportradar.com><br/>1 Javier Acosta <javier.acosta@beeonline.es><br/>1 Andreas Steinmetz <ast@domdv.de><br/>1 Yaniv Simhi <yaniv.simhi@gmail.com><br/>1 Taylor Hawkes <th71852@gmail.com><br/>1 Paddy Grice <paddy@wizaner.com><br/>1 Panos Gkikakis <roeften@gmail.com><br/>1 David Cunningham<br/>1 Olle Johansson<br/>1 Gareth Palmer <gareth@acsdata.co.nz><br/>1 Jared Biel <jared.biel@bolderthinking.com><br/>1 Jeffrey C. Ollie <jeff@ocjtech.us><br/>1 Bryant Zimmerman<br/>1 Ronald Raikes<br/>1 Karsten Wemheuer<br/>1 Sean Bright<br/>1 Jeff Collell<br/>1 Tove Hjelm <tj.direct@thenordicvoice.com><br/>1 Warren Selby <wcselby@selbytech.com><br/>1 klaus3000 <ramon@pernau.at><br/>1 Josh Kitchens <jkitchens@microcom.tv><br/>1 Dmitry Melekhov<br/>1 Sergio Medina Toledo <lumasepa@gmail.com><br/>1 dant <d@tucny.com><br/>1 Stephan Eisvogel <eisvogel@embinet.de><br/>1 David Justl<br/>1 Steven T. Wheeler<br/>1 warren smith<br/>1 Juergen Spies <Juergen.Spies@vivai.de><br/>1 Frank DiGennaro <fsd@voipbusiness.us><br/>1 Ben Langfeld<br/>1 Kinsey Moore<br/>1 Brad Latus<br/>1 Dan Jenkins <dan@nimbleape.com><br/>1 Barry Chern<br/>1 Max Man<br/>1 Eelco Brolman <e.brolman@telecats.nl><br/>1 Guido Falsi <madpilot@freebsd.org><br/>1 Dmitry Melekhov <dm@belkam.com><br/>1 viniciusfontes <vinicius@canall.com.br><br/>1 Krzysztof Trempala <k.trempala@slican.pl><br/>1 LEI FU <lei.fu@modulis.ca><br/>1 Gianluca Merlo <gianluca.merlo@gmail.com><br/>1 David Brillert<br/>1 John Zhong<br/>1 Thomas Airmont<br/>1 Marcello Ceschia<br/>1 vadim <vadim@mbdsys.com><br/>1 Daniel Flounders <trade@pcserv.org.uk><br/>1 Nick Ruggles<br/>1 Andrey Biglari<br/>1 Arveno Santoro <a.santoro@ecoricerche.it><br/>1 David Cunningham <dcunningham@voisonics.com><br/>1 Ben Klang <bklang@mojolingo.com><br/>1 Olle Johansson <oej@edvina.net><br/>1 Ben Merrills<br/>1 Jeremy Kister<br/>1 Timo Teräs <timo.teras@iki.fi><br/>1 jeffrey putnam<br/>1 Eelco Brolman<br/>1 Aleksei Kulakov <each.nir.vine@gmail.com><br/>1 Bojan Nemčić<br/>1 Nir Simionovich<br/>1 Artem Volodin<br/>1 Christoph Timm <christoph.timm@voipfuture.com><br/>1 John Kiniston <johnkiniston@gmail.com><br/>1 Kristian Hogh<br/>1 Private Name<br/>1 Ronald Raikes <reraikes@avweb.com><br/>1 Yaniv Simhi<br/>1 Artem Volodin <rus.diezel@gmail.com><br/>1 Ed Hynan <edhynan@gmail.com><br/>1 Lorne Gaetz<br/>1 feyfre <panych.y@gmail.com><br/>1 Jared Biel<br/>1 Andrew Zherdin <andrew.zherdin@binastar.de><br/>1 Guido Falsi<br/>1 Mitch Claborn<br/>1 Richard Miller <rich@ndpcci.com><br/>1 Ben Klang<br/>1 Olivier Krief <olivier.krief@gmail.com><br/>1 Peter Katzmann <peter.katzmann@edag.de><br/>1 Marcello Ceschia <marcello.ceschia@gmx.net><br/>1 Ed Hynan<br/>1 Dan Tucny<br/>1 Jacques Peacock <jpeacock@callconnection.com><br/>1 Denis Alberto Martinez <dmartinez@digium.com><br/>1 Mateusz Kowalski<br/>1 Mark Petersen <asterisk.org@zombie.dk><br/>1 Carlos Oliva <carlos.oliva@invoxcontact.com><br/>1 Josh Colp<br/>1 HZMI8gkCvPpom0tM<br/>1 Thomas Thompson<br/>1 sungtae kim <pchero21@gmail.com><br/>1 Stefan27 (on IRC)<br/>1 jeffrey putnam <jputnam@telesign.com><br/>1 Tony Ching <tony@u2systems.com><br/>1 Walter Doekes<br/>1 Dwayne Hubbard <dhubbard@digium.com><br/>1 HZMI8gkCvPpom0tM <fuxfwgc4a2i1gr@gmail.com><br/>1 PowerPBX <canuck15@hotmail.com><br/>1 Frederic Van Espen <frederic.ve@gmail.com><br/>1 Damian Ivereigh<br/>1 Sebastian Kemper<br/>1 mdu113 <mulitskiy@acedsl.com><br/>1 Justin T. Gibbs <gibbs@scsiguy.org><br/>1 Alexandre Fournier <alexandre.fournier@kiplink.fr><br/>1 Ben Langfeld <ben@langfeld.me><br/>1 George Ladoff <georgeladoff@gmail.com><br/>1 Juergen Spies<br/>1 Max Man <afterme@gmail.com><br/>1 Gareth Palmer<br/>1 Jeffrey Walton <noloader@gmail.com><br/>1 Patric Marschall <patric.marschall@1und1.de><br/>1 Andrew Zherdin<br/>1 Martin Moučka <moucka.m@gmail.com><br/>1 Yura Kocyuba<br/>1 Mark Petersen<br/>1 Gareth Blades<br/>1 Filip Frank <Frenkfil@email.cz><br/>1 Guenther Kelleter<br/>1 LEI FU<br/>1 Matthias Urlichs <smurf@smurf.noris.de><br/>1 Marco Paland <info@paland.com><br/>1 Ilya Trikoz <jleed@me.com><br/>1 Elazar Broad<br/>1 Alec Davis <sivad.a@paradise.net.nz><br/>1 Jaco Kroon<br/>1 Barry Chern <bchurl@columbus.rr.com><br/>1 dea <dan_austin@fitawi.com><br/>1 Dan Jenkins<br/>1 Michael Keuter<br/>1 Lorne Gaetz <lgaetz@gmail.com><br/>1 Matt Hoskins<br/>1 Stephan Eisvogel<br/>1 Luke Hulsey<br/>1 Ben Smithurst<br/>1 David Justl <david@rockauto.com><br/>1 Tyler Cambron <tcambron@digium.com><br/>1 Patric Marschall<br/>1 Mateusz Kowalski <mateusz.kowalski@cern.ch><br/>1 Dave Cabot <dave.cabot@ossengineering.com><br/>1 Krzysztof Trempala<br/>1 Tove Hjelm<br/>1 Frederic Van Espen<br/>1 Malcolm Davenport <malcolmd@digium.com><br/>1 Private Name <sales@minixel.com><br/>1 John Kiniston<br/>1 Stefan Engström<br/>1 Maciej Szmigiero<br/>1 Steven T. Wheeler <swheeler@usinternet.com><br/>1 Simon Arlott <issues.asterisk.simon@arlott.org><br/>1 PSDK <hyavari26@gmail.com><br/>1 Hiroaki Komatsu <komatsu.hiroaki@po.ntts.co.jp><br/>1 Atis Lezdins <a.lezdinsh@ctncorp.com><br/>1 Marco Paland<br/>1 Malcolm Davenport<br/>1 Jeff Collell <auctionsjeff@gmail.com><br/>1 Timo Teräs<br/>1 Ilya Trikoz<br/>1 Anatoli <me@anatoli.ws><br/>1 Martin Moučka<br/>1 Gergely Dömsödi <doome@uhusystems.com><br/>1 Simon Arlott<br/>1 Richard Miller<br/>1 Kevin Scott Adams <ksatllc@att.net><br/>1 Aleksandr Gordeev <axonaro@gmail.com><br/>1 Dudás József <jozsef.dudas@gmail.com><br/>1 Nic Colledge <nic@njcolledge.net><br/>1 Kristian Høgh <kfh@uni-tel.dk><br/>1 Paddy Grice<br/>1 Peter Whisker <peter@whisker.org.uk><br/>1 Elazar Broad <elazar@thebroadfamily.com><br/>1 Matthias Urlichs<br/>1 xrobau <rob@wpm4L.com><br/>1 Janusz Karolak <janusz_1942@op.pl><br/>1 Atis Lezdins<br/>1 Anthony Messina <amessina@messinet.com><br/>1 cervajs <cervajs@fpf.slu.cz><br/>1 Filip Jenicek <phill@janevim.cz><br/>1 abelbeck <lonnie@abelbeck.com><br/>1 Peter Whisker<br/>1 Anatoli <me@anatoli.ws><br/>1 Osaulenko Alexander <a.osaulenko@callway.com.ua><br/>1 Bojan Nemčić <bojan.nemcic@voxdiversa.hr><br/>1 Gergely Dömsödi<br/>1 Torrey Searle <tsearle@gmail.com><br/>1 not here <anonymous@anonymous.invalid><br/>1 Dave Cabot<br/>1 Christoph Timm<br/>1 Ben Merrills <ben@xdev.net><br/></td></tr>
</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>New Feature</h3><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25480">ASTERISK-25480</a>: [patch]Add field PauseReason on QueueMemberStatus<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e13719bff1c4a723edf08252da17fef04b6f88cf">[e13719bff1]</a> Rodrigo Ramírez Norambuena -- app_queue: Added reason pause of member</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25670">ASTERISK-25670</a>: Add regcontext to PJSIP<br/>Reported by: Daniel Journo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=22801a06ee26993db9ad26b39b83315de29410cb">[22801a06ee]</a> Daniel Journo -- pjsip: Add option global/regcontext</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25259">ASTERISK-25259</a>: chan_pjsip: Add rtptimeout support<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27497217912a82ca243a9d5e9acfbbb597faf323">[2749721791]</a> Joshua Colp -- pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.</li>
</ul><br><h4>Category: Channels/chan_sip/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17899">ASTERISK-17899</a>: Handle crypto lifetime in SDES-SRTP negotiation<br/>Reported by: Dwayne Hubbard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd8ac00f2410d72408be4e68ad518a85d6373343">[dd8ac00f24]</a> Olle Johansson -- channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25419">ASTERISK-25419</a>: Dialplan Application for Integration of StatsD<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e0040b88f83688e67d71521177cf4fa962bf32a">[1e0040b88f]</a> Tyler Cambron -- StatsD: Add res_statsd compatibility</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25252">ASTERISK-25252</a>: ARI: Add the ability to manipulate log channels<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df9ce3636695781be6ab2479f90766a56747dbd7">[df9ce36366]</a> Scott Emidy -- ARI: Retrieve existing log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e9f1bc08cbda7759707c30b8883b266555d0fefc">[e9f1bc08cb]</a> Scott Emidy -- ARI: Creating log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78364132ce94d9ded24ae6e6ab44b97d256b506d">[78364132ce]</a> Scott Emidy -- ARI: Deleting log channels</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ae762634c317fbcbd98a8c34d2474f7d4b654ed">[1ae762634c]</a> Benjamin Ford -- ARI: Rotate log channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25238">ASTERISK-25238</a>: ARI: Support push configuration<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bcf6d2801d1b1ed7073ab560bdbe3d0047b1b2c">[8bcf6d2801]</a> Matt Jordan -- ARI: Add support for push configuration of dynamic object</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb76b88bafebd69bba31d85acd24fa5d46f3d59a">[bb76b88baf]</a> Matt Jordan -- main/sorcery: Don't fail object set creation from JSON if field fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f13c2226a35f465ab70c8e25a885f1f3cdaa1c5">[5f13c2226a]</a> Matt Jordan -- main/format_cap: Parse capabilities generated by ast_format_cap_get_names</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25173">ASTERISK-25173</a>: ARI: Add the ability to load/reload/unload an Asterisk module<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3384e64ef62b0f61dec9f4b4f345b6db74348ae3">[3384e64ef6]</a> Benjamin Ford -- ARI: Fixed unload mode for unload module.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1aafadf8148a7cf66f73beeb6fe711d98b678fc5">[1aafadf814]</a> Benjamin Ford -- ARI: Added new functionality to reload a single module.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9dcae23cfceedece83568d2194df00ca62f7d53c">[9dcae23cfc]</a> Benjamin Ford -- ARI: Added new functionality to unload a single module.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c219a98d2b46a61996518fd2791b7bb4437969fb">[c219a98d2b]</a> Benjamin Ford -- ARI: Added new functionality to load a single module.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73e35d20deb57281874939f553fea9fdced2e260">[73e35d20de]</a> Benjamin Ford -- ARI: Added new functionality to get information on a single module.</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24922">ASTERISK-24922</a>: ARI: Add the ability to intercept hold and raise an event<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72cbb6df55bad972bf12800946f3c0b219aca049">[72cbb6df55]</a> Matt Jordan -- funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab803ec3422e5377b5aa4769fbcc6b315d167d8b">[ab803ec342]</a> Matt Jordan -- ARI: Add the ability to intercept hold and raise an event</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24703">ASTERISK-24703</a>: ARI: Add the ability to "transfer" (redirect) a channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1995baad71412b740743f2be82d7323f7c15de12">[1995baad71]</a> Matt Jordan -- ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24919">ASTERISK-24919</a>: res_pjsip_config_wizard: Ability to write contents to file<br/>Reported by: Ray Crumrine<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e848dae7bb8f0f36691856635e1a6518ade85bf">[5e848dae7b]</a> gtjoseph -- res_pjsip_config_wizard: Add command to export primitive objects</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25377">ASTERISK-25377</a>: res_pjsip: Change default "From user" from UUID to something more palatable<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac62928d6b7333a0d502be2eba99c238549ae1a3">[ac62928d6b]</a> Mark Michelson -- res_pjsip: Change default from user value.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25259">ASTERISK-25259</a>: chan_pjsip: Add rtptimeout support<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27497217912a82ca243a9d5e9acfbbb597faf323">[2749721791]</a> Joshua Colp -- pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25419">ASTERISK-25419</a>: Dialplan Application for Integration of StatsD<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e0040b88f83688e67d71521177cf4fa962bf32a">[1e0040b88f]</a> Tyler Cambron -- StatsD: Add res_statsd compatibility</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25640">ASTERISK-25640</a>: pbx: Deadlock on features reload and state change hint.<br/>Reported by: Krzysztof Trempala<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1600ebca7d9a233f39ae56e8d662ba8d95a067c1">[1600ebca7d]</a> Kevin Harwell -- pbx: Deadlock between contexts container and context_merge locks</li>
</ul><br><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25227">ASTERISK-25227</a>: No audio at in-band announcements in ooh323 channel<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=71408df2b82c932329d250a6077475e4f51a2b0d">[71408df2b8]</a> Alexander Anikin -- chan_ooh323: Add ProgressIndicator IE with inband info available</li>
</ul><br><h4>Category: Addons/res_config_mysql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18252">ASTERISK-18252</a>: queue_log mysql time column data format<br/>Reported by: Gareth Blades<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e67e8d5c7ff2ca83a9ec8a32513f4963971a0001">[e67e8d5c7f]</a> Alexandre Fournier -- res_config_mysql: Fix broken column type checking</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25041">ASTERISK-25041</a>: [patch]Broken column type checking in res_config_mysql addon<br/>Reported by: Alexandre Fournier<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e67e8d5c7ff2ca83a9ec8a32513f4963971a0001">[e67e8d5c7f]</a> Alexandre Fournier -- res_config_mysql: Fix broken column type checking</li>
</ul><br><h4>Category: Applications/app_agent_pool</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24737">ASTERISK-24737</a>: When agent not logged in, agent status shows unavailable, queue status shows agent invalid<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e2832b35b53c36531d35fd2c36b654d8f4d697">[05e2832b35]</a> Richard Mudgett -- app_agent_pool: Fix initial module load agent device state reporting.</li>
</ul><br><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19470">ASTERISK-19470</a>: Documentation on app_amd is incorrect<br/>Reported by: Frank DiGennaro<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a4c27baf47eeeebf3a4e6660065ee9b7c5a0ee31">[a4c27baf47]</a> Matt Jordan -- apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation</li>
</ul><br><h4>Category: Applications/app_chanspy</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25321">ASTERISK-25321</a>: [patch]DeadLock ChanSpy with call over Local channel<br/>Reported by: Filip Frank<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=336cae73cc85bac570028ec72dac42fa75691e75">[336cae73cc]</a> Walter Doekes -- app_chanspy: Fix occasional deadlock with ChanSpy and Local channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25247">ASTERISK-25247</a>: choppy audio when spying on a g722 channel, chan_sip or chan_pjsip<br/>Reported by: hristo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1493f900e76304ecf28eed30d37fba8e54d926b">[f1493f900e]</a> Joshua Colp -- audiohook: Read the correct number of samples based on audiohook format.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24828">ASTERISK-24828</a>: Fix Frame Leaks<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=169058e73fbe0b7cb7ff7e57c457093e5296532b">[169058e73f]</a> Kevin Harwell -- app_chanspy, channel: fix frame leaks</li>
</ul><br><h4>Category: Applications/app_confbridge</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20987">ASTERISK-20987</a>: non-admin users, who join muted conference are not being muted<br/>Reported by: hristo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d0abf86e798c61a5eff83cdfa3ed38154e0fe3e">[1d0abf86e7]</a> Richard Mudgett -- app_confbridge: Add ability to get the muted conference state.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e51e5c7fd1f747ee67ec830cf3a78f54102e12b">[3e51e5c7fd]</a> Richard Mudgett -- app_confbridge: Make non-admin users join a muted conference muted.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25253">ASTERISK-25253</a>: confbridge volume options and other volume controls such as func_volume don't work<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f509730cb93875ba0a78835fd38b8dbd1cdff3f7">[f509730cb9]</a> Joshua Colp -- audiohook: Use manipulated frame instead of dropping it.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24749">ASTERISK-24749</a>: ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge<br/>Reported by: Philippe Bolduc<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7e5056b393734508eeef02fa95c25f8de05ac733">[7e5056b393]</a> Kevin Harwell -- app_confbridge: Default the template option to a compatible default profile.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24841">ASTERISK-24841</a>: ConfBridge: Strange sampling rates chosen when channels have multiple native formats<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=82bc0fd3ade77394e13062b6097732c224e77eaa">[82bc0fd3ad]</a> Richard Mudgett -- res_fax: Fix latent bug exposed by ASTERISK-24841 changes.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13cd99682d5eb8511f72cad9a4995faa2ac77048">[13cd99682d]</a> Richard Mudgett -- chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4441bb6a2560d7781d87b15eaa2cef58f7645da8">[4441bb6a25]</a> Richard Mudgett -- Bridging: Eliminate the unnecessary make channel compatible with bridge operation.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9cdadc168cbc3da709e02c9faf3315e0a5507ba1">[9cdadc168c]</a> Matt Jordan -- res/res_pjsip_sdp_rtp: Revert portion of r432195</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24864">ASTERISK-24864</a>: app_confbridge: file playback blocks dtmf<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fd434a210f4255b741ed90dd0634b5a1a8ca241a">[fd434a210f]</a> Kevin Harwell -- app_confbridge: file playback blocks dtmf</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24719">ASTERISK-24719</a>: ConfBridge recording channels get stuck when recording started/stopped more than once<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eda125f98dd1a11f41b7a89f96fbe3fb754c8963">[eda125f98d]</a> Richard Mudgett -- app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24723">ASTERISK-24723</a>: confbridge: CLI command 'confbridge list XXXX' no longer displays user menus<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1a176937898ad54eab130b719f7dc20eb0df5b85">[1a17693789]</a> Matt Jordan -- app_confbridge: Restore user's menu name to CLI output of 'confbridge list'</li>
</ul><br><h4>Category: Applications/app_dial</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24958">ASTERISK-24958</a>: Forwarding loop detection inhibits certain desirable scenarios<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7debb986a51937886392f7f444f29483528f94ec">[7debb986a5]</a> Alec Davis -- app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f1a8dbe9241cf967000d9f5f359830948da3c9a">[4f1a8dbe92]</a> Mark Michelson -- Detect potential forwarding loops based on count.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25212">ASTERISK-25212</a>: [patch]Segfault when using DEBUG_FD_LEAKS<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6551e16e03cc8e172d9ad8f75a040750a29a5e6e">[6551e16e03]</a> Walter Doekes -- astfd: Fix buffer overflow in DEBUG_FD_LEAKS.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0d018e249177b755cafc9cf8d1fbd6e53839df7">[f0d018e249]</a> Joshua Colp -- res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24682">ASTERISK-24682</a>: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=894d4d781cd361a87ffc0ed343b47345bbfeaaaf">[894d4d781c]</a> Matt Jordan -- apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values</li>
</ul><br><h4>Category: Applications/app_directory</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25087">ASTERISK-25087</a>: Asterisk segfault when using Directory application with alias option and specific mailbox configuration<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a2f4d03c87bb67892a8b846c59bd26e9163054e9">[a2f4d03c87]</a> Richard Mudgett -- app_directory: Fix crash when using the alias option 'a'.</li>
</ul><br><h4>Category: Applications/app_meetme</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25569">ASTERISK-25569</a>: app_meetme: Audio quality issues<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddf4dddf4ffa4866806f5ae6d5bc6cfaa9aa6fdb">[ddf4dddf4f]</a> Corey Farrell -- app_meetme: Set default value for audio_buffers.</li>
</ul><br><h4>Category: Applications/app_mixmonitor</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25322">ASTERISK-25322</a>: Crash occurs when using MixMonitor with t() or r() options.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3a56bee83c01dbb538620a11947d420b17cf458">[b3a56bee83]</a> Richard Mudgett -- audiohook.c: Fix MixMonitor crash when using the r() or t() options.</li>
</ul><br><h4>Category: Applications/app_page</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25384">ASTERISK-25384</a>: Regular Asterisk crashes when using Page application. "user_data is NULL"<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f15cd93f0b2cb622d54061515b815e3ebbe76b1">[5f15cd93f0]</a> Richard Mudgett -- app_page.c: Fix crash when forwarding with a predial handler.</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25800">ASTERISK-25800</a>: [patch] Calculate talktime when is first call answered<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79dc5e2f00e0b78f6278c34740b824eb289d1518">[79dc5e2f00]</a> Rodrigo Ramírez Norambuena -- app_queue: fix Calculate talktime when is first call answered</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25442">ASTERISK-25442</a>: using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c) <br/>Reported by: Carlos Oliva<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae428d846006771c45518301e5987cdbabec6e79">[ae428d8460]</a> Carlos Oliva -- app_queue: update RT members when the 1st call joins a queue with no agents</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25561">ASTERISK-25561</a>: app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!<br/>Reported by: Alec Davis<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7debb986a51937886392f7f444f29483528f94ec">[7debb986a5]</a> Alec Davis -- app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked!</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25423">ASTERISK-25423</a>: Caller gets no Connected line update during call pickup.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b1e7583c150cc9bafcb567c789b6c23c60e2c71">[6b1e7583c1]</a> Richard Mudgett -- app_queue.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6bf304bf251e31ab7c7a4d89508445b84fb5d551">[6bf304bf25]</a> Richard Mudgett -- app_queue.c: Factor out a connected line update routine.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e36b5f1e8e1fd1e805184fca015bb0808b5e7fb8">[e36b5f1e8e]</a> Richard Mudgett -- app_dial.c: Make 'A' option pass COLP updates.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=747bfac895c4d51f55ce687322aa6a95c52be4e2">[747bfac895]</a> Richard Mudgett -- app_dial.c: Force COLP update if outgoing channel name changed.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14481d9aa0365682ace22c97d5c115166be5429d">[14481d9aa0]</a> Richard Mudgett -- app_dial.c: Factor out a connected line update routine.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25399">ASTERISK-25399</a>: app_queue: AgentComplete event has wrong reason<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4fb95bbc4e4928dd3403a20d401c285a568f0d09">[4fb95bbc4e]</a> Kevin Harwell -- app_queue: AgentComplete event has wrong reason</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25185">ASTERISK-25185</a>: Segfault in app_queue on transfer scenarios<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6409e7b11a2310196a9978b30a6b79e2760be592">[6409e7b11a]</a> Kevin Harwell -- app_queue: Crash when transferring</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25215">ASTERISK-25215</a>: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember<br/>Reported by: Lorne Gaetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e5f5b9f384eef0389a7e35d40c91d3586869a125">[e5f5b9f384]</a> Richard Mudgett -- app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25038">ASTERISK-25038</a>: Queue log "EXITWITHTIMEOUT" does not always contain waiting time<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87263b47b51bf727e04f352f988acb508924e423">[87263b47b5]</a> Ivan Poddubny -- app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23319">ASTERISK-23319</a>: Segmentation fault in queue_exec at app_queue.c<br/>Reported by: Vadim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e05c8ae68e95367ea60fdc5e8a5f2cf7f24e0f32">[e05c8ae68e]</a> Stefan Engström -- apps/app_queue: Prevent possible crash when evaluating queue penalty rules</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24267">ASTERISK-24267</a>: Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel<br/>Reported by: Mitch Claborn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1a0979d437ea809a1c642977c0b6b109e2a8246a">[1a0979d437]</a> Kevin Harwell -- app_queue: Update sample conf documenation</li>
</ul><br><h4>Category: Applications/app_record</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25410">ASTERISK-25410</a>: app_record: RECORDED_FILE variable not being populated<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeddee39fb492ea5ec873bdf02ea3858c5282601">[aeddee39fb]</a> Kevin Harwell -- app_record: RECORDED_FILE variable not being populated</li>
</ul><br><h4>Category: Applications/app_transfer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24015">ASTERISK-24015</a>: app_transfer fails with PJSIP channels<br/>Reported by: Private Name<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1995baad71412b740743f2be82d7323f7c15de12">[1995baad71]</a> Matt Jordan -- ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25082">ASTERISK-25082</a>: Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d49d64b79c365841c01aff1b25be43af8065bd2d">[d49d64b79c]</a> Jonathan Rose -- app_voicemail: fix moving when old messages full</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24626">ASTERISK-24626</a>: Voicemail passwords not being stored in ARA<br/>Reported by: Paddy Grice<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1fc823c770461689d44a1a8bd14f57a6ddf92372">[1fc823c770]</a> Matt Jordan -- dynamic realtime: Updates fail to work due to update fields being passed over</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24709">ASTERISK-24709</a>: [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event<br/>Reported by: Gareth Palmer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=beb20440e0db725e7b0a2768574c60525e44af12">[beb20440e0]</a> Gareth Palmer -- apps/app_voicemail: Trigger MWI notification with MixMonitor m() option</li>
</ul><br><h4>Category: Applications/app_voicemail/IMAP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24786">ASTERISK-24786</a>: [patch] - Asterisk terminates when playing a voicemail stored in LDAP<br/>Reported by: Graham Barnett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=055001716cd6548ec644f3cc62841b3b7ff4f66c">[055001716c]</a> Graham Barnett -- app_voicemail: Fix crash with IMAP backends when greetings aren't present</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24787">ASTERISK-24787</a>: [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime<br/>Reported by: Graham Barnett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c7bdf62a956c39fff738f581e03d78920e9a0981">[c7bdf62a95]</a> Graham Barnett -- apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24288">ASTERISK-24288</a>: [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup<br/>Reported by: LEI FU<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e659b3e53d63287e486f1f4224b8f05d3475dc3f">[e659b3e53d]</a> LEI FU -- app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend</li>
</ul><br><h4>Category: Applications/app_voicemail/ODBC</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24288">ASTERISK-24288</a>: [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup<br/>Reported by: LEI FU<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e659b3e53d63287e486f1f4224b8f05d3475dc3f">[e659b3e53d]</a> LEI FU -- app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend</li>
</ul><br><h4>Category: Bridges/bridge_holding</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25271">ASTERISK-25271</a>: Parking & blind transfer: Transferer channel not hung up if no MOH<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8458b8d441c2f4143ff135163ff3da4f88fe14c8">[8458b8d441]</a> Jonathan Rose -- holding_bridge: ensure moh participants get frames</li>
</ul><br><h4>Category: Bridges/bridge_native_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25240">ASTERISK-25240</a>: bridge_native_rtp: Direct media wrongfully started when completing attended transfer<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d558b00c8503a002bc8f0173fd1e5f911fc6483e">[d558b00c85]</a> Joshua Colp -- bridge_native_rtp.c: Don't start native RTP bridging after attended transfer.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25171">ASTERISK-25171</a>: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4a2ef9e4ef27488609bb01fc55e965cd93a9ad5">[e4a2ef9e4e]</a> Joshua Colp -- channel: Remove ignore of answer on non-outgoing channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24459">ASTERISK-24459</a>: bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible<br/>Reported by: Yaniv Simhi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4fad85f9bf58ed2a11acd5559ca9bce097f2f1cf">[4fad85f9bf]</a> Kevin Harwell -- res_pjsip_sdp_rtp: wrong bridge chosen when the DTMF mode is not compatible</li>
</ul><br><h4>Category: Bridges/bridge_simple</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><br><h4>Category: Bridges/bridge_softmix</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24797">ASTERISK-24797</a>: bridge_softmix: G.729 codec license held<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b73246a9dc2592fd01fbe993f5be07f01250c17">[5b73246a9d]</a> Kevin Harwell -- bridge_softmix: G.729 codec license held</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24344">ASTERISK-24344</a>: CDR_PROP(disable) disables CDR only for first dialed party<br/>Reported by: Janusz Karolak<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de8c7f46ed0c1212054b6b6cfd33663549ebd94c">[de8c7f46ed]</a> Matt Jordan -- main/cdr: Carry over the disable flag when 'disable all' is specified</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24443">ASTERISK-24443</a>: CDR fields (dst, dcontext) empty in transfer call started from Macro<br/>Reported by: Arveno Santoro<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78ea356e78a4dc7c88b2212d1c4bf700bc5c5701">[78ea356e78]</a> Matt Jordan -- main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25090">ASTERISK-25090</a>: CLI core show channel truncates cdr variables<br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=091b436007c55fbcc827cceb97a414a172aa36af">[091b436007]</a> snuffy -- cdr: Fix 'core show channel' CDR variable truncation.</li>
</ul><br><h4>Category: CDR/cdr_custom</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25179">ASTERISK-25179</a>: CDR(billsec,f) and CDR(duration,f) report incorrect values<br/>Reported by: Gianluca Merlo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46b2de55f95927b67f9309426353f2f5f5d7c830">[46b2de55f9]</a> Matt Jordan -- funcs/func_cdr: Correctly report high precision values for duration and billsec</li>
</ul><br><h4>Category: CDR/cdr_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24976">ASTERISK-24976</a>: cdr_odbc not include new columns added on 1.8<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f611fa0e8fd54d191a4c2104b5d1ed4b1f8da97">[7f611fa0e8]</a> Rodrigo Ramírez Norambuena -- cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5dd43856e88a80ef9a0ed050c88ff1316929761">[d5dd43856e]</a> Rodrigo Ramírez Norambuena -- cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version</li>
</ul><br><h4>Category: CDR/cdr_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24959">ASTERISK-24959</a>: [patch]CLI command cdr show pgsql status<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07e729cc7b79b2aa7f3da1f034ef75bf5f882c5e">[07e729cc7b]</a> Rodrigo Ramírez Norambuena -- cdr_pgsql: Fix CLI "cdr show pgsql status" command.</li>
</ul><br><h4>Category: CEL/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25647">ASTERISK-25647</a>: bug of cel_radius.c: wrong point of ADD_VENDOR_CODE<br/>Reported by: Aaron An<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86eae38d7ecf64dc3c737a8050c3a7699f2be626">[86eae38d7e]</a> Aaron An -- cel/cel_radius: Fix wrong pointer.</li>
</ul><br><h4>Category: Channels/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25025">ASTERISK-25025</a>: Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=181ae3b8d929bd798a98b584868d1eac85b2eee9">[181ae3b8d9]</a> Joshua Colp -- stasis: Fix dial masquerade datastore lifetime</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=077979618b5a472c4bd3612680f71275a18d3841">[077979618b]</a> Mark Michelson -- Prevent potential crash on blond transfer.</li>
</ul><br><h4>Category: Channels/chan_dahdi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25315">ASTERISK-25315</a>: DAHDI channels send shortened duration DTMF tones.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=256bc52b6684141b100c5018dbf0ad3ce6111585">[256bc52b66]</a> Richard Mudgett -- chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=800e0ea48de88c016e1f477edf7db7b9aadc4b54">[800e0ea48d]</a> Richard Mudgett -- chan_dahdi.c: Lock private struct for ast_write().</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25257">ASTERISK-25257</a>: [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope<br/>Reported by: Patric Marschall<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=abb14ac5b8d833176560716199521aa260dc2d0e">[abb14ac5b8]</a> Patric Marschall -- sig_pri.h: force_restart_unavailable_chans in wrong scope</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21893">ASTERISK-21893</a>: Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c<br/>Reported by: Aleksandr Gordeev<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c780b6e431438fb837832e311c585ba2cb573fec">[c780b6e431]</a> Richard Mudgett -- chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25034">ASTERISK-25034</a>: chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=904f5d98f6b011d12f5a8f07641f539d29e10d38">[904f5d98f6]</a> Richard Mudgett -- chan_dahdi: Improve force_restart_unavailable_chans option description.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3c310a28c801d4d64d4d1763c9c53f6f18a2388">[d3c310a28c]</a> Richard Mudgett -- chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19608">ASTERISK-19608</a>: Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.<br/>Reported by: Denis Alberto Martinez<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d3c310a28c801d4d64d4d1763c9c53f6f18a2388">[d3c310a28c]</a> Richard Mudgett -- chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24895">ASTERISK-24895</a>: After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.<br/>Reported by: Andrew Zherdin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d08446ec36600348090eb2451c5a9a7aadd633de">[d08446ec36]</a> Richard Mudgett -- chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24869">ASTERISK-24869</a>: Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel<br/>Reported by: viniciusfontes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b56c1914fa54817e88be92ed8394aa6f64aa26ad">[b56c1914fa]</a> Kevin Harwell -- bridge.c: NULL app causes crash during attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24825">ASTERISK-24825</a>: Caller ID not recognized using Centrex/Distinctive dialing<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8cced7767c09c7300f47c2cc445227f0f488b899">[8cced7767c]</a> Richard Mudgett -- chan_dahdi/sig_analog: Fix distinctive ring detection to suck less.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17588">ASTERISK-17588</a>: Caller ID on TDM410P *UK* PSTN<br/>Reported by: Daniel Flounders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8cced7767c09c7300f47c2cc445227f0f488b899">[8cced7767c]</a> Richard Mudgett -- chan_dahdi/sig_analog: Fix distinctive ring detection to suck less.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24689">ASTERISK-24689</a>: Segfault on hangup after outgoing PRI-Euroisdn call<br/>Reported by: Marcel Manz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40547e7210d5ca2ba651bbcdb7af50eb330a5ed8">[40547e7210]</a> Richard Mudgett -- ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association.</li>
</ul><br><h4>Category: Channels/chan_iax2</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24983">ASTERISK-24983</a>: IAX deadlock between hangup and scheduled actions (ex. largrq)<br/>Reported by: Y Ateya<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf98c744d5bb7e5f015f201bc79c58462f7aaaed">[cf98c744d5]</a> Yousf Ateya -- chan_iax2: Prevent deadlock between hangup and sending lagrq/ping</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22352">ASTERISK-22352</a>: [patch] IAX2 custom qualify timer is not taken into account<br/>Reported by: Frederic Van Espen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c39faa4729a0c4490352fa5707e064de17068aff">[c39faa4729]</a> Y Ateya -- channels/chan_iax2: Improve POKE expiration time calculation for lossy networks</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24894">ASTERISK-24894</a>: [patch] iax2_poke_noanswer expiration timer too short<br/>Reported by: Y Ateya<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c39faa4729a0c4490352fa5707e064de17068aff">[c39faa4729]</a> Y Ateya -- channels/chan_iax2: Improve POKE expiration time calculation for lossy networks</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21211">ASTERISK-21211</a>: chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault<br/>Reported by: Jaco Kroon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05397ad01e92069296a68e12bad72f89bfff6c14">[05397ad01e]</a> Jaco Kroon -- chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno]</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24451">ASTERISK-24451</a>: chan_iax2: reference leak in sched_delay_remove<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb9448a1ae18bf16074346bdaae5e0d29d3bcd38">[eb9448a1ae]</a> Corey Farrell -- Create work around for scheduler leaks during shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24600">ASTERISK-24600</a>: Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock<br/>Reported by: Jeff Collell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6af6a216a1fcfb1dcefbfa6dc657d3781f3a24d9">[6af6a216a1]</a> Richard Mudgett -- CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge.</li>
</ul><br><h4>Category: Channels/chan_local</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25250">ASTERISK-25250</a>: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f63552052710d2b0a0f33b8fd93dd00083f74b74">[f635520527]</a> Mark Michelson -- Local channels: Alternate solution to ringback problem.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=54b25c80c8387aea9eb20f9f4f077486cbdf3e5d">[54b25c80c8]</a> Mark Michelson -- Local channels: Do not block control -1 payloads.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24267">ASTERISK-24267</a>: Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel<br/>Reported by: Mitch Claborn<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1a0979d437ea809a1c642977c0b6b109e2a8246a">[1a0979d437]</a> Kevin Harwell -- app_queue: Update sample conf documenation</li>
</ul><br><h4>Category: Channels/chan_mgcp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25220">ASTERISK-25220</a>: [patch]Closing of fd -1 in chan_mgcp.c<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5a262be78113f58d54e45da69225d1ee856a51e">[a5a262be78]</a> Walter Doekes -- chan_mgcp: Don't call close on fd -1.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25849">ASTERISK-25849</a>: chan_pjsip: transfers with direct media sometimes drops audio<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a40520fe9c4b5a00d8de425b18163cc5aefe8d0">[6a40520fe9]</a> Kevin Harwell -- chan_pjsip: ref leak when checking direct_media_glare</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9444ddadf8525d1ce66a1faf1db97f9f6c265ca4">[9444ddadf8]</a> Kevin Harwell -- chan_pjsip: transfers with direct media reinvite has wrong address/port</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32fc784284b570a05841d95c6d9a373b4bf3a35d">[32fc784284]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25637">ASTERISK-25637</a>: Multi homed server using wrong IP<br/>Reported by: Daniel Journo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=219c204a418cbc82ca529837de53cb332ada6b37">[219c204a41]</a> gtjoseph -- pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25675">ASTERISK-25675</a>: Endpoint not listed as Unreachable<br/>Reported by: Daniel Journo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b10fc917340bbbfe6222fa7c6131f004912879a">[4b10fc9173]</a> gtjoseph -- Revert "pjsip_location: Delete contact_status object when contact is deleted"</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24779">ASTERISK-24779</a>: Passthrough OPUS codec not working with chan_pjsip<br/>Reported by: PowerPBX<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=33752e0837122fa73551dd3c424765477455b433">[33752e0837]</a> Sean Bright -- res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25404">ASTERISK-25404</a>: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c<br/>Reported by: Chet Stevens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=426263a64dc5dc80a51b6950ee0cb6b46f5f052c">[426263a64d]</a> Richard Mudgett -- chan_pjsip: Fix crash on reINVITE before initial INVITE completes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25258">ASTERISK-25258</a>: chan_pjsip: Incorrect format switch on received RTP packet<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c626ceb64e11f7d18b97023f09072a992060121">[2c626ceb64]</a> Joshua Colp -- chan_pjsip: Don't change formats when frame of unsupported format is received.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25183">ASTERISK-25183</a>: PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=653f2087e0d2edc9df8a0154c09e2e608e13a5c5">[653f2087e0]</a> Richard Mudgett -- res_pjsip_session.c: Fix crash on call disconnect.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ada7346792452f021911063668997f79fdabc1f1">[ada7346792]</a> Richard Mudgett -- res_pjsip: Need to use the same serializer for a pjproject SIP transaction.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25091">ASTERISK-25091</a>: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge<br/>Reported by: Ilya Trikoz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9015bb4c8c9882de35066c6586189ab78268a12f">[9015bb4c8c]</a> Mark Michelson -- Resolve race conditions involving Stasis bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25156">ASTERISK-25156</a>: chan_pjsip’s CHAN_START cel event lacks the correct context and exten<br/>Reported by: cloos<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=30a0f2d9acd0f7c14013d830a0b4bf673d0af2d0">[30a0f2d9ac]</a> Matt Jordan -- chan_pjsip: Set the context and extension on the channel when created</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24996">ASTERISK-24996</a>: chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3278fe5327c6d34966ef6c87191d8d7d4fae4f66">[3278fe5327]</a> Ashley Sanders -- chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25018">ASTERISK-25018</a>: pjsip show endpoints crashes asterisk when qualified aors present<br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99fb87ae13a1fc28bd5043600def3b373f85a0f6">[99fb87ae13]</a> gtjoseph -- res_pjsip: Fix SEGV on pending-qualify contacts</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24845">ASTERISK-24845</a>: pjsip send notify not working with Cisco phone<br/>Reported by: Carl Fortin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1da9ec969ddfbc9a1450e698af20913df92e5fd9">[1da9ec969d]</a> Mark Michelson -- res_pjsip_outbound_authenticator: Increase CSeq on authed requests.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24933">ASTERISK-24933</a>: T38 fails negotiation<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f21b45db4920debe736b710951b4860fb420f306">[f21b45db49]</a> Jonathan Rose -- res_pjsip_t38: Fix FAX failures when using PJSIP with authentication</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24781">ASTERISK-24781</a>: PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1e9552b087c68d412d6233034a1ad6e4eda02bd">[b1e9552b08]</a> Richard Mudgett -- chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24771">ASTERISK-24771</a>: ${CHANNEL(pjsip)} - segfault<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5d2623675818231aaeed804c803e3c94addeff3f">[5d26236758]</a> Joshua Colp -- chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24666">ASTERISK-24666</a>: Security Vulnerability: RTP not closed after sip call using unsupported codec<br/>Reported by: Y Ateya<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c068fc096c7a7f2c80e30b6a89dc057902201e0">[8c068fc096]</a> Mark Michelson -- Fix file descriptor leak in RTP code.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1582929f9c27330ac58420e2329421a4713b70c">[c1582929f9]</a> Mark Michelson -- Prevent possible race condition on dual redirect of channels in the same bridge.</li>
</ul><br><h4>Category: Channels/chan_sip/CodecHandling</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25160">ASTERISK-25160</a>: [patch] Opus Codec: SIP/SDP line fmtp missing when called internally<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d8d3991390bcae96dd5e33c59dcfe17e9932b7d4">[d8d3991390]</a> Alexander Traud -- format: Register format-attribute module with cached formats.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24543">ASTERISK-24543</a>: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs<br/>Reported by: Taylor Hawkes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1256aedf66b062c011959c422df1fa08e9f55522">[1256aedf66]</a> Alexander Traud -- chan_sip: Do not send all codecs on INVITE.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25484">ASTERISK-25484</a>: [patch] autoframing=yes has no effect<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=869ef2a8ee4e4df271227d6b9b48470e44ad4831">[869ef2a8ee]</a> Alexander Traud -- chan_sip: Fix autoframing=yes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25309">ASTERISK-25309</a>: [patch] iLBC 20 advertised<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f68c995bc97c9b6cb4887043b344087d82aeef10">[f68c995bc9]</a> Alexander Traud -- chan_sip: Fix negotiation of iLBC 30.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25182">ASTERISK-25182</a>: [patch] on CLI sip reload, new codecs get appended only<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a419c69def639745ac9988b3800501f68dfef350">[a419c69def]</a> Alexander Traud -- chan_sip: Reload peer without its old capabilities.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21777">ASTERISK-21777</a>: Asterisk tries to transcode video instead of audio<br/>Reported by: Nick Ruggles<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a63f7ad04a00bfe8e6b47fd1a60de6691813c0d1">[a63f7ad04a]</a> Richard Mudgett -- translate.c: Only select audio codecs to determine the best translation choice.</li>
</ul><br><h4>Category: Channels/chan_sip/DatabaseSupport</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24772">ASTERISK-24772</a>: ODBC error in realtime sippeers when device unregisters under MariaDB<br/>Reported by: Richard Miller<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=29f3ff0b615bb0847727e2df45d5590f5f9ebcee">[29f3ff0b61]</a> Richard Miller -- channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB</li>
</ul><br><h4>Category: Channels/chan_sip/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25023">ASTERISK-25023</a>: Deadlock in chan_sip in update_provisional_keepalive<br/>Reported by: Arnd Schmitter<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de04308ae434bf3910725dc7d2c5a66ecd12d963">[de04308ae4]</a> Richard Mudgett -- chan_sip.c: Fix mwi resub deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f6627a8a4f2ef68759cea0c27e7aaec25ac031c">[5f6627a8a4]</a> Richard Mudgett -- chan_sip.c: Fix registration timeout and expire deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32bd7a64f98d5e9cf176cba1c701145201b1f987">[32bd7a64f9]</a> Richard Mudgett -- chan_sip.c: Fix t38id deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43556b800b7838fc989f243a2974d33a8f8a12ce">[43556b800b]</a> Richard Mudgett -- chan_sip.c: Fix reinviteid deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=38c1cdab2ce3f5f00cd5a6a0f561910d9265bea7">[38c1cdab2c]</a> Richard Mudgett -- chan_sip.c: Fix packet retransid deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4ad55c888bc6c7e0d382e716d1d8ed4219d6cb2">[e4ad55c888]</a> Richard Mudgett -- chan_sip.c: Fix waitid deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98d5669c280dbf5ffcd8a0431aa44e7f465fcefe">[98d5669c28]</a> Richard Mudgett -- chan_sip.c: Fix session timers deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9cb8f73226126db70bac54fd7af8093ab05ffd6f">[9cb8f73226]</a> Richard Mudgett -- chan_sip.c: Fix autokillid deadlock potential.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c5c7f48a15a17fcc29ff022223df5229ef9a003f">[c5c7f48a15]</a> Richard Mudgett -- chan_sip.c: Fix provisional_keepalive_sched_id deadlock.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f959d84dfddb68532299a7669419f0b78286a75f">[f959d84dfd]</a> Richard Mudgett -- chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f3225ddccb3bdf3673cec8f2d453be9ceb98029">[5f3225ddcc]</a> Richard Mudgett -- chan_sip.c: Clear scheduled immediate events on unload.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a74971771d9bf302cb26238996eeb3cb1df7af8">[7a74971771]</a> Richard Mudgett -- sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b2d29064454e1118fe987954242d91deef752aff">[b2d2906445]</a> Richard Mudgett -- sched.c: Ensure oldest expiring entry runs first.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25397">ASTERISK-25397</a>: [patch]chan_sip: File descriptor leak with non-default timert1<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c81a052c88270a2bef6d4641559cf22837b31a6">[3c81a052c8]</a> Richard Mudgett -- AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25364">ASTERISK-25364</a>: [patch]Issue a TCP connection(kernel) and thread of asterisk is not released<br/>Reported by: Hiroaki Komatsu<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14b41115e363766633aec67f67e9764521b74f5c">[14b41115e3]</a> Jonathan Rose -- chan_sip: Add TCP/TLS keepalive to TCP/TLS server</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25610">ASTERISK-25610</a>: Asterisk crash during "sip reload"<br/>Reported by: Dudás József<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b992014dcf8f1d343e95a06868d4ebd14619d33">[2b992014dc]</a> Richard Mudgett -- chan_sip: Fix crash involving the bogus peer during sip reload.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25476">ASTERISK-25476</a>: chan_sip loses registrations after a while<br/>Reported by: Michael Keuter<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7c88e11aa753e046fe58a19ac82320c81cc6e2b">[e7c88e11aa]</a> Richard Mudgett -- sched.c: Make not return a sched id of 0.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4aed349a7bd2e62d82c5e9535f7cf69263eeb60a">[4aed349a7b]</a> Richard Mudgett -- Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions)</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d9156d10f8941f1da90bf81109904432a2f293d">[6d9156d10f]</a> Richard Mudgett -- Audit improper usage of scheduler exposed by 5c713fdf18f.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07583c288828a496cd7730b55112128fea31eaef">[07583c2888]</a> Steve Davies -- Further fixes to improper usage of scheduler</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24543">ASTERISK-24543</a>: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs<br/>Reported by: Taylor Hawkes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1256aedf66b062c011959c422df1fa08e9f55522">[1256aedf66]</a> Alexander Traud -- chan_sip: Do not send all codecs on INVITE.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25346">ASTERISK-25346</a>: chan_sip: Overwriting answered elsewhere hangup cause on call pickup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c01111223f9dbd383a4dd1cf786b63eff214f238">[c01111223f]</a> Joshua Colp -- chan_sip: Allow call pickup to set the hangup cause.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25250">ASTERISK-25250</a>: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f63552052710d2b0a0f33b8fd93dd00083f74b74">[f635520527]</a> Mark Michelson -- Local channels: Alternate solution to ringback problem.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=54b25c80c8387aea9eb20f9f4f077486cbdf3e5d">[54b25c80c8]</a> Mark Michelson -- Local channels: Do not block control -1 payloads.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22805">ASTERISK-22805</a>: res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP <br/>Reported by: Dmitry Burilov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25212">ASTERISK-25212</a>: [patch]Segfault when using DEBUG_FD_LEAKS<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6551e16e03cc8e172d9ad8f75a040750a29a5e6e">[6551e16e03]</a> Walter Doekes -- astfd: Fix buffer overflow in DEBUG_FD_LEAKS.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25202">ASTERISK-25202</a>: Hints extension state broken between 13.3.2 and 13.4<br/>Reported by: cervajs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=399cd8bcd9e53f30d4a36b67200281407f27798e">[399cd8bcd9]</a> Matt Jordan -- main/pbx: Resolve case sensitivity regression in PBX hints</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25171">ASTERISK-25171</a>: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4a2ef9e4ef27488609bb01fc55e965cd93a9ad5">[e4a2ef9e4e]</a> Joshua Colp -- channel: Remove ignore of answer on non-outgoing channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25163">ASTERISK-25163</a>: Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74616ae43d4e24e914ee612846a464da5b241b9b">[74616ae43d]</a> Joshua Colp -- chan_sip: Destroy peers without holding peers container lock.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24835">ASTERISK-24835</a>: Early Media Not working with Chan SIP and Asterisk 13<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af458e2e601d156618b2a39c6bb09f1b82766f43">[af458e2e60]</a> Kevin Harwell -- chan_sip: make progressinband default to no</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24882">ASTERISK-24882</a>: chan_sip: Improve usage of REF_DEBUG<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=958bc84cafd43963f0a016f8979ec962c49af517">[958bc84caf]</a> Corey Farrell -- chan_sip: Simplify dialog/peer references, improve REF_DEBUG output.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24876">ASTERISK-24876</a>: Investigate reference leaks from tests/channels/local/local_optimize_away<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fddae99dd8f3f77245a5d9811734259a5f9e138">[7fddae99dd]</a> Corey Farrell -- chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24838">ASTERISK-24838</a>: chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13e715b30c1e4fff81d6c42896a5ef5ade29b74a">[13e715b30c]</a> Richard Mudgett -- chan_sip: Fix realtime locking inversion when poking a just built peer.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21845">ASTERISK-21845</a>: maxcalls exceeded, Asterisk sends out 480 and also BYE<br/>Reported by: Tony Ching<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34989bd9c8adf79af2aaf7a2fb1965cb590f36c5">[34989bd9c8]</a> Makoto Dei -- channels/chan_sip: Don't send a BYE after final response when PBX thread fails</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-15434">ASTERISK-15434</a>: [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller<br/>Reported by: Makoto Dei<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34989bd9c8adf79af2aaf7a2fb1965cb590f36c5">[34989bd9c8]</a> Makoto Dei -- channels/chan_sip: Don't send a BYE after final response when PBX thread fails</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23214">ASTERISK-23214</a>: chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddff640f949a2fc47520fb52ed86a00282113df3">[ddff640f94]</a> Matt Jordan -- channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24800">ASTERISK-24800</a>: Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill<br/>Reported by: JoshE<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=978649a56838c7502a5869b93c697a7c66e5921e">[978649a568]</a> Matt Jordan -- channels/chan_sip: Fix crash when transmitting packet after thread shutdown</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22436">ASTERISK-22436</a>: [patch] No BYE to masqueraded channel on INVITE with replaces<br/>Reported by: Eelco Brolman<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03445a147e8e38486f49e0a117b640e32236009d">[03445a147e]</a> Jeremiah Gowdy -- Blocked revisions 431620</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24355">ASTERISK-24355</a>: [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'<br/>Reported by: HZMI8gkCvPpom0tM<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9210648bbe116b9c19b7c1a6304a2dd00ff0cca3">[9210648bbe]</a> HZMI8gkCvPpom0tM -- chan_sip: Case insensitive comparison of "defaultuser" parameter.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24628">ASTERISK-24628</a>: [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)<br/>Reported by: Karsten Wemheuer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9735a13429ea8a03c99e963fd5d4044b909d1dde">[9735a13429]</a> Karsten Wemheuer -- chan_sip: Send CANCEL via original INVITE destination even after UPDATE request</li>
</ul><br><h4>Category: Channels/chan_sip/IPv6</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25443">ASTERISK-25443</a>: [patch]IPv6 - Potential issue in via header parsing<br/>Reported by: ffs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f939e2bd48598d721aa18af8182b7b6b91a9fe95">[f939e2bd48]</a> Florian Sauerteig -- chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25100">ASTERISK-25100</a>: asterisk coredump if host has an IPv6 address that end with ::80<br/>Reported by: Mark Petersen<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97a6ce1717bd0c4b1b4305f10f13fd5ec9bb7441">[97a6ce1717]</a> Ivan Poddubny -- Astobj2: Correctly treat hash_fn returning INT_MIN</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18032">ASTERISK-18032</a>: [patch] - IPv6 and IPv4 NAT not working<br/>Reported by: Christoph Timm<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be13c72142d9bf8a7a22308ad2e4142c819775af">[be13c72142]</a> Valentin Vidić -- chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled</li>
</ul><br><h4>Category: Channels/chan_sip/Interoperability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25135">ASTERISK-25135</a>: [patch]RTP Timeout hangup cause code missing<br/>Reported by: Olle Johansson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f8707ae9a57b47a742c051e6714416f46b156118">[f8707ae9a5]</a> Olle Johansson -- channels/chan_sip: Set cause code to 44 on RTP timeout</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25396">ASTERISK-25396</a>: chan_sip: Extremely long callerid name causes invalid SIP<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b59c4d82b58a7a10e1791f9ed5759af8ac637df2">[b59c4d82b5]</a> Walter Doekes -- chan_sip: Fix From header truncation for extremely long CALLERID(name).</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25154">ASTERISK-25154</a>: [patch]fromtag may need to be updated after successful call dialog match<br/>Reported by: Damian Ivereigh<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f57f3f8ec63752ccd1d87d7c6737f64043cb8a9">[3f57f3f8ec]</a> Damian Ivereigh -- chan_sip.c: Update dialog fromtag after request with auth</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24646">ASTERISK-24646</a>: PJSIP changeset 4899 breaks TLS<br/>Reported by: Stephan Eisvogel<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=22fc3359dabe8b0f6dfaa4073a9380a9fcd31ec7">[22fc3359da]</a> Mark Michelson -- Use SIPS URIs in Contact headers when appropriate.</li>
</ul><br><h4>Category: Channels/chan_sip/Registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24715">ASTERISK-24715</a>: chan_sip: stale nonce causes failure<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e2b493b8f0d04b5f881cf748001c3ef4b272d53a">[e2b493b8f0]</a> Kevin Harwell -- chan_sip: stale nonce causes failure</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24673">ASTERISK-24673</a>: outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74a13629e210e00e79b430184c4197856877afe0">[74a13629e2]</a> Matt Jordan -- channels/chan_sip: Fix registration leak during reload</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24640">ASTERISK-24640</a>: Registration pending stays forever after sip reload<br/>Reported by: Max Man<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74a13629e210e00e79b430184c4197856877afe0">[74a13629e2]</a> Matt Jordan -- channels/chan_sip: Fix registration leak during reload</li>
</ul><br><h4>Category: Channels/chan_sip/SRTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24550">ASTERISK-24550</a>: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake<br/>Reported by: Osaulenko Alexander<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24887">ASTERISK-24887</a>: [patch]tags in a=crypto lines do not accept 2 or more digits<br/>Reported by: Makoto Dei<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17d6ede337439fe84e4f9efa7b65eb4ceaace0c1">[17d6ede337]</a> Corey Edwards -- main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17721">ASTERISK-17721</a>: Incoming SRTP calls that specify a key lifetime fail<br/>Reported by: Terry Wilson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd8ac00f2410d72408be4e68ad518a85d6373343">[dd8ac00f24]</a> Olle Johansson -- channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20233">ASTERISK-20233</a>: SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"<br/>Reported by: tootai<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd8ac00f2410d72408be4e68ad518a85d6373343">[dd8ac00f24]</a> Olle Johansson -- channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22748">ASTERISK-22748</a>: SRTP Crypto Offer With Lifetime Not Accepted<br/>Reported by: Alejandro Mejia<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd8ac00f2410d72408be4e68ad518a85d6373343">[dd8ac00f24]</a> Olle Johansson -- channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI</li>
</ul><br><h4>Category: Channels/chan_sip/Security Framework</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25722">ASTERISK-25722</a>: ASAN & testsute: stack-buffer-overflow in sip_sipredirect<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6823bb0c4f33dd19c11cd30f23650a194c2dbb7">[a6823bb0c4]</a> Corey Farrell -- chan_sip: Fix buffer overrun in sip_sipredirect.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25320">ASTERISK-25320</a>: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25af2d71c863062868b8bda6cf83515a1935d27e">[25af2d71c8]</a> Kevin Harwell -- chan_sip.c: wrong peer searched in sip_report_security_event</li>
</ul><br><h4>Category: Channels/chan_sip/T.38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25609">ASTERISK-25609</a>: [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)<br/>Reported by: Filip Jenicek<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=142d4fefb8db0ac2c30b18f75dc415093fb77f27">[142d4fefb8]</a> Filip Jenicek -- chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c)</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24449">ASTERISK-24449</a>: Reinvite for T.38 UDPTL fails if SRTP is enabled<br/>Reported by: Andreas Steinmetz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f26d4618eb7dff4e6c8e8dfd31765bb331bdf2ba">[f26d4618eb]</a> Andreas Steinmetz -- chan_sip: Allow T.38 switch-over when SRTP is in use.</li>
</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24847">ASTERISK-24847</a>: [security] [patch] tcptls: certificate CN NULL byte prefix bug<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7674409061bcd5be8fa0d59a7f53221057d441a">[f767440906]</a> Maciej Szmigiero -- Security/tcptls: MitM Attack potential from certificate with NULL byte in CN.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22748">ASTERISK-22748</a>: SRTP Crypto Offer With Lifetime Not Accepted<br/>Reported by: Alejandro Mejia<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd8ac00f2410d72408be4e68ad518a85d6373343">[dd8ac00f24]</a> Olle Johansson -- channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24799">ASTERISK-24799</a>: [patch] make fails with undefined reference to SSLv3_client_method<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=374013d8177687229e59584b48506c0f83ff02aa">[374013d817]</a> Alexander Traud -- tcptls: Handle new OpenSSL compile time option to disable SSLv3</li>
</ul><br><h4>Category: Channels/chan_sip/Transfers</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25226">ASTERISK-25226</a>: chan_sip: Channel leak in branch 13 on early replaces call pickup<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0f565663b49e14c15d7b5e6e9ff7396956b91f6">[e0f565663b]</a> Walter Doekes -- chan_sip: Fix early call pickup channel leak.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24628">ASTERISK-24628</a>: [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)<br/>Reported by: Karsten Wemheuer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9735a13429ea8a03c99e963fd5d4044b909d1dde">[9735a13429]</a> Karsten Wemheuer -- chan_sip: Send CANCEL via original INVITE destination even after UPDATE request</li>
</ul><br><h4>Category: Channels/chan_sip/WebSocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25317">ASTERISK-25317</a>: asterisk sends too many stun requests<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d228b62fd437e02c0638684c1f44c92e5f1e3948">[d228b62fd4]</a> gtjoseph -- stasis_cache_pattern: Backport to 13</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">ASTERISK-24146</a>: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec<br/>Reported by: Aleksei Kulakov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28d924307940700ce2321572b016fdd8263ac7ad">[28d9243079]</a> Eugene Voityuk -- chan_sip.c: Start ICE negotiation when response is sent or received.</li>
</ul><br><h4>Category: Channels/chan_skinny</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25494">ASTERISK-25494</a>: build: GCC 5.1.x catches some new const, array bounds and missing paren issues<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f593e7c38b260a9769d0e01b1edf24098599cd7">[5f593e7c38]</a> gtjoseph -- build: GCC 5.1.x catches some new const, array bounds and missing paren issues</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-13721">ASTERISK-13721</a>: memory leak in "strings.c"<br/>Reported by: pj<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=222fbe1d9ad76897e1c91faefff53379f45c3983">[222fbe1d9a]</a> Corey Farrell -- Build System: Replace comment about setting menuselect defaults.</li>
</ul><br><h4>Category: Channels/chan_unistim</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><br><h4>Category: Codecs/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25616">ASTERISK-25616</a>: Warning with a Codec Module which supports PLC with FEC<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=69e3d40ad74883db9bb9b34d6aed71a536e8cf3c">[69e3d40ad7]</a> Alexander Traud -- translate: Avoid a warning message when doing FEC within Opus Codec.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25498">ASTERISK-25498</a>: Asterisk crashes when negotiating g729 without that module installed<br/>Reported by: Ben Langfeld<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=072d94183ce2b6d2272543732dd5d47390099bb3">[072d94183c]</a> Jonathan Rose -- Fix crash in audiohook translate to slin</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25353">ASTERISK-25353</a>: [patch] Transcoding while different in Frame size = Frames lost<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b88c54fa4bd537bde46519abb95e30a5f96673ac">[b88c54fa4b]</a> Alexander Traud -- translate: Fix transcoding while different in frame size.</li>
</ul><br><h4>Category: Codecs/codec_adpcm</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24717">ASTERISK-24717</a>: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=888bb49618cdf952a00481ea148c0be462593f29">[888bb49618]</a> Ivan Poddubny -- Fix buffer overflow in slin sample frames generation.</li>
</ul><br><h4>Category: Codecs/codec_gsm</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24717">ASTERISK-24717</a>: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=888bb49618cdf952a00481ea148c0be462593f29">[888bb49618]</a> Ivan Poddubny -- Fix buffer overflow in slin sample frames generation.</li>
</ul><br><h4>Category: Codecs/codec_ilbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24717">ASTERISK-24717</a>: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=888bb49618cdf952a00481ea148c0be462593f29">[888bb49618]</a> Ivan Poddubny -- Fix buffer overflow in slin sample frames generation.</li>
</ul><br><h4>Category: Codecs/codec_lpc10</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24717">ASTERISK-24717</a>: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=888bb49618cdf952a00481ea148c0be462593f29">[888bb49618]</a> Ivan Poddubny -- Fix buffer overflow in slin sample frames generation.</li>
</ul><br><h4>Category: Codecs/codec_resample</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25599">ASTERISK-25599</a>: [patch] SLIN Resampling Codec only 80 msec<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=876600ce6e1b165dd068f30c763e5c517c3b6ae8">[876600ce6e]</a> Alexander Traud -- codec_resample: Increase buffer for Opus Codec with FEC.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b76c196e13c08f022f99defc13ec98f0942c2fec">[b76c196e13]</a> Alexander Traud -- codec_resample: Increase buffer for Opus Codec.</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25113">ASTERISK-25113</a>: install_prereq in Debian 8 without "standard system utilities"<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=efcf9a96db6c40ea658f597714d0ff841ec0d412">[efcf9a96db]</a> Rodrigo Ramírez Norambuena -- install_prereq: Check if is installed aptitude otherwise to install.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24632">ASTERISK-24632</a>: install_prereq script installs pjproject without IPv6 support<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cbe63ab2830b0cb18f61109baebd004bdde32598">[cbe63ab283]</a> Joshua Colp -- install_prereq: Tweak flags when configuring pjproject.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24048">ASTERISK-24048</a>: [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts<br/>Reported by: Ben Klang<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=072db5e1b9201b3d7219a12ed559a9e856d535b6">[072db5e1b9]</a> Ben Klang -- contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hosts</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24474">ASTERISK-24474</a>: sip_to_pjsip.py lacks documentation and does not function<br/>Reported by: John Kiniston<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4791d629d1885d7c908a2ab9b85f0e8de966de23">[4791d629d1]</a> Scott Griepentrog -- sip_to_pjsip: improve ability to parse input files</li>
</ul><br><h4>Category: Core/AstDB</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25400">ASTERISK-25400</a>: Hints broken when "CustomPresence" doesn't exist in AstDB<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3357678b949ebbc2f7aa00144d19bdbcfb1896b1">[3357678b94]</a> Ivan Poddubny -- func_presencestate: Return "not_set" when no data is set in AstDB</li>
</ul><br><h4>Category: Core/AstMM</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25048">ASTERISK-25048</a>: Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5875bf183c255f19ededdb1004f35fd15cd1e6cf">[5875bf183c]</a> Corey Farrell -- Astobj2: Fix initialization order of refdebug and AO2_DEBUG.</li>
</ul><br><h4>Category: Core/Bridging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25771">ASTERISK-25771</a>: ARI:Crash - Attended transfers of channels into Stasis application.<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40d9e9e2384fcf1da715d7a34680f02631c8ec64">[40d9e9e238]</a> Kevin Harwell -- bridge.c: Crash during attended transfer when missing a local channel half</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff3da61c35f44761516e29f315cefd908ad895d8">[ff3da61c35]</a> Kevin Harwell -- res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25600">ASTERISK-25600</a>: bridging: Inconsistency in BRIDGEPEER<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eadad24b595c3b6e5f0472f9936e7e37259308b5">[eadad24b59]</a> Jonathan Rose -- Unset BRIDGEPEER when leaving a bridge</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25341">ASTERISK-25341</a>: bridge: Hangups may get lost when executing actions<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c2dab1e888e59cb429ed61219815bd00eee66c0">[6c2dab1e88]</a> Joshua Colp -- bridge: Kick channel from bridge if hung up during action.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25250">ASTERISK-25250</a>: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f63552052710d2b0a0f33b8fd93dd00083f74b74">[f635520527]</a> Mark Michelson -- Local channels: Alternate solution to ringback problem.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=54b25c80c8387aea9eb20f9f4f077486cbdf3e5d">[54b25c80c8]</a> Mark Michelson -- Local channels: Do not block control -1 payloads.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24782">ASTERISK-24782</a>: StasisEnd event not present for channel that was swapped out for another after completing attended transfer<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97ee0ee6c6d5f37591183339999d8cb936bf517a">[97ee0ee6c6]</a> Kevin Harwell -- bridge.c: Fixed race condition during attended transfer</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=35a99b639474f9140fc294c184bb8f0afb1936cf">[35a99b6394]</a> Kevin Harwell -- bridge.c: Hangup attended transfer target if bridged</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d754f70239921561449884e85bd9794b5f515cd9">[d754f70239]</a> Kevin Harwell -- bridge.c: Hangup attended transfer target after it has been swapped out</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25157">ASTERISK-25157</a>: bridging: Performing a blonde transfer does not result in connected line updates<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbb067279e7d7555c5090546572a0d01f796fe55">[dbb067279e]</a> Joshua Colp -- bridge: When performing a blonde transfer update connected line information.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24869">ASTERISK-24869</a>: Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel<br/>Reported by: viniciusfontes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b56c1914fa54817e88be92ed8394aa6f64aa26ad">[b56c1914fa]</a> Kevin Harwell -- bridge.c: NULL app causes crash during attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24752">ASTERISK-24752</a>: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=feddab7944a12746bcbdb6fc82cc9d4951d61eb5">[feddab7944]</a> Richard Mudgett -- HTTP: Stop accepting requests on final system shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24539">ASTERISK-24539</a>: Compile fails on OSX because of sem_timedwait in bridge_channel.c<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=702d79de2a2341954ef599f4c5a4c5f31c1b5edc">[702d79de2a]</a> David M. Lee -- Various fixes for OS X</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1582929f9c27330ac58420e2329421a4713b70c">[c1582929f9]</a> Mark Michelson -- Prevent possible race condition on dual redirect of channels in the same bridge.</li>
</ul><br><h4>Category: Core/Bridging/bridge_basic</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25696">ASTERISK-25696</a>: bridge_basic: don't cache xferfailsound during a transfer<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=660fedecb7fb723b5fcab8aa9e3f29c0c788e988">[660fedecb7]</a> Kevin Harwell -- bridge_basic: don't cache xferfailsound during an attended transfer</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25697">ASTERISK-25697</a>: bridge_basic: don't play an attended transfer fail sound after target hangs up<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=83feb7db3b33674b75fbe2259f0841e731d74bf3">[83feb7db3b]</a> Kevin Harwell -- bridge_basic: don't play an attended transfer fail sound after target hangs up</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25641">ASTERISK-25641</a>: bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel<br/>Reported by: Dmitry Melekhov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3024cad103b6b661e0f179ca61f43738f8e09e3">[b3024cad10]</a> Richard Mudgett -- bridge_basic.c: Fix GOTO_ON_BLINDXFR</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24513">ASTERISK-24513</a>: Local channel apparently leaked in off-nominal DTMF attended transfer<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=566907fabd6acfd20f59145044649991af23d976">[566907fabd]</a> Scott Griepentrog -- bridge: avoid leaking channel during blond transfer pt2</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a99df47c09e9d9f51ea6afba76b259832eadb51">[6a99df47c0]</a> Scott Griepentrog -- bridge: avoid leaking channel during blond transfer</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25730">ASTERISK-25730</a>: build: make uninstall after make distclean tries to remove root<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aee8448bc24d6da80dda577cfc14b6c9055577a0">[aee8448bc2]</a> gtjoseph -- build_system: Prevent goals needing makeopts from running when it's missing</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25434">ASTERISK-25434</a>: Compiler flags not reported in 'core show settings' despite usage during compilation<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d098d00424a3c7ae2c2b2b26ce31d0889c506478">[d098d00424]</a> Corey Farrell -- Fix cli display of build options.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25074">ASTERISK-25074</a>: Regression: Recent clang-related change broke cross compiling of Asterisk<br/>Reported by: Sebastian Kemper<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6627de830b7a121939662a640e7b0aaa3899f61b">[6627de830b]</a> Sebastian Kemper -- General: Fix recent menuselect-related cross compile regression</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25027">ASTERISK-25027</a>: Build System: Many ARI modules are missing dependencies.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=366ea63438cb698d11d95462bafde843ad5a262e">[366ea63438]</a> Corey Farrell -- res_ari_bridges: Add missing dependencies.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d61f03c4f90d9d1dea979dc758cc13fe78d2c789">[d61f03c4f9]</a> Corey Farrell -- ARI: Fix missing dependencies.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e4624ad21bc3eb7cedd6c7e41145ebc7cb6d86f">[3e4624ad21]</a> Corey Farrell -- res_pjsip: Remove incorrect MODULEINFO from presence_xml.c.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fed9faab8d4c1d237a862a60a35a13a5ccb3f61f">[fed9faab8d]</a> Corey Farrell -- Git Migration: Create doc/rest-api when needed.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25028">ASTERISK-25028</a>: Build System: Unneeded defines in asterisk/buildopts.h<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac1f0090eb5f93f8d37f512a94aa6b16affb77e1">[ac1f0090eb]</a> Corey Farrell -- Build System: Prevent unneeded changes to asterisk/buildopts.h.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24954">ASTERISK-24954</a>: Git migration: Asterisk version numbers are incompatible with the Test Suite<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e996d8f728bd03ca8ade8067c71d5e8a91c74fe4">[e996d8f728]</a> Matt Jordan -- build_tools/make_version: Update version parsing for Git migration</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24932">ASTERISK-24932</a>: Asterisk 13.x does not build with GCC 5.0<br/>Reported by: Jeffrey C. Ollie<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=95de71f2475f49ab8d0c7e9d93d5147847e0c010">[95de71f247]</a> gtjoseph -- build: Fixes for gcc 5 compilation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24880">ASTERISK-24880</a>: [patch]Compilation under OpenBSD <br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08a88aab159b99588bf705e3e325813c07ae5d0f">[08a88aab15]</a> snuffy -- Fix compilation issues for OpenBSD</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20399">ASTERISK-20399</a>: Compilation on some systems requires the -fnested-functions flag<br/>Reported by: David M. Lee<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5bc032567ac1af93e8d5320a7a6c126a8639611">[f5bc032567]</a> Diederik de Groot -- Add support for the clang compiler; update RAII_VAR to use BlocksRuntime</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20850">ASTERISK-20850</a>: [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.<br/>Reported by: Diederik de Groot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5bc032567ac1af93e8d5320a7a6c126a8639611">[f5bc032567]</a> Diederik de Groot -- Add support for the clang compiler; update RAII_VAR to use BlocksRuntime</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18105">ASTERISK-18105</a>: most of asterisk modules are unbuildable in cygwin environment<br/>Reported by: feyfre<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=de86b30dbaaa616ca1752690c89eaa1bea4798a2">[de86b30dba]</a> Matt Jordan -- make: Remove 'res_features' from libraries to link against with cygwin/mingw32</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24544">ASTERISK-24544</a>: Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=702d79de2a2341954ef599f4c5a4c5f31c1b5edc">[702d79de2a]</a> David M. Lee -- Various fixes for OS X</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23991">ASTERISK-23991</a>: [patch]asterisk.pc file contains a small error in the CFlags returned<br/>Reported by: Diederik de Groot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89a431df8475df407a9e305e4cd8a16e6d231e93">[89a431df84]</a> Diederik de Groot -- build_tools/mkpkgconfig: Fix Cflags concatenation error in asterisk.pc</li>
</ul><br><h4>Category: Core/CallCompletionSupplementaryServices</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24142">ASTERISK-24142</a>: CCSS: crash during shutdown due to device lookup in destroyed container<br/>Reported by: David Brillert<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6adf26f14dbe2ed0addd5d07910c3b5546005987">[6adf26f14d]</a> Corey Farrell -- Replace most uses of ast_register_atexit with ast_register_cleanup.</li>
</ul><br><h4>Category: Core/Channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25690">ASTERISK-25690</a>: Hanging up when executing connected line sub does not cause hangup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32b29d7b027b43f78ffda3b85adb35653d5365b3">[32b29d7b02]</a> Joshua Colp -- app: Queue hangup if channel is hung up during sub or macro execution.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24991">ASTERISK-24991</a>: Check for ao2_alloc failure in __ast_channel_internal_alloc<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad1a118632098a5d2d746d157f4c6f3d15694f65">[ad1a118632]</a> Corey Farrell -- Check for ao2_alloc failure in __ast_channel_internal_alloc.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24380">ASTERISK-24380</a>: core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a63f7ad04a00bfe8e6b47fd1a60de6691813c0d1">[a63f7ad04a]</a> Richard Mudgett -- translate.c: Only select audio codecs to determine the best translation choice.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21038">ASTERISK-21038</a>: Bad command completion of "core set debug channel"<br/>Reported by: Richard Kenner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=287a22435f7063e997312f954c635b85a8afdbe8">[287a22435f]</a> Joshua Colp -- core: Fix tab completion of "core set debug channel" CLI command.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24828">ASTERISK-24828</a>: Fix Frame Leaks<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=169058e73fbe0b7cb7ff7e57c457093e5296532b">[169058e73f]</a> Kevin Harwell -- app_chanspy, channel: fix frame leaks</li>
</ul><br><h4>Category: Core/CodecInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25172">ASTERISK-25172</a>: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e99e654d75a2428ce4b8bc504acf2ec1927779ed">[e99e654d75]</a> Joshua Colp -- app_dial: Hold reference to calling channel formats when dialing outbound.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21777">ASTERISK-21777</a>: Asterisk tries to transcode video instead of audio<br/>Reported by: Nick Ruggles<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a63f7ad04a00bfe8e6b47fd1a60de6691813c0d1">[a63f7ad04a]</a> Richard Mudgett -- translate.c: Only select audio codecs to determine the best translation choice.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-16779">ASTERISK-16779</a>: Cannot disallow unknown format ''<br/>Reported by: Atis Lezdins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c03a5f2e792635a3e61886df94e52802ef40371">[5c03a5f2e7]</a> Matt Jordan -- main/frame: Don't report empty disallow values as an error</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24796">ASTERISK-24796</a>: Codecs and bucket schema's prevent module unload<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=93c9c3af2f63f7f25dc24220066f2e667761e78f">[93c9c3af2f]</a> Corey Farrell -- Allow shutdown to unload modules that register bucket scheme's or codec's.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=439e6e1c5d2eaecb4ea132e2c11bb13c42201ab4">[439e6e1c5d]</a> Joshua Colp -- media: Fix crash when determining sample count of a frame during shutdown.</li>
</ul><br><h4>Category: Core/Configuration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25725">ASTERISK-25725</a>: core: Incorrect XML documentation may result in weird behavior<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f22074e5d9ed1882be976299311b8e093d25e1da">[f22074e5d9]</a> Joshua Colp -- config: Allow options to register when documentation is unavailable.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25700">ASTERISK-25700</a>: main/config: Clean config maps on shutdown.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f5f30cf825baad5e2aa08a875c5e7d9b084be65">[3f5f30cf82]</a> Corey Farrell -- main/config: Clean config maps on shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25683">ASTERISK-25683</a>: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG <br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=935d641f3bb71d786e69ff99a6b6b99165fd4ca2">[935d641f3b]</a> Mark Michelson -- Remove res/ari/* content during 'make clean'.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25042">ASTERISK-25042</a>: asterisk.conf options override command-line options.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3cdb7950f05dbfe0b180a778c48f99f97345958c">[3cdb7950f0]</a> Corey Farrell -- Fix processing of asterisk.conf debug=yes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24231">ASTERISK-24231</a>: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1fc823c770461689d44a1a8bd14f57a6ddf92372">[1fc823c770]</a> Matt Jordan -- dynamic realtime: Updates fail to work due to update fields being passed over</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23733">ASTERISK-23733</a>: 'reload acl' fails if acl.conf is not present on startup<br/>Reported by: Richard Kenner<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f88460115fab0fb7171e3da23d9418579d198fa4">[f88460115f]</a> Joshua Colp -- acl: Fix reloading of configuration if configuration file does not exist at startup.</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25681">ASTERISK-25681</a>: devicestate: Engine thread is not shut down<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fc3dad96531cdf1d484044e24240517c19c4342">[0fc3dad965]</a> Corey Farrell -- devicestate: Cleanup engine thread during graceful shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25601">ASTERISK-25601</a>: json: Audit reference usage and thread safety<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a9d6fc571d08de45ac3b9cfb78db9007f7b8ed48">[a9d6fc571d]</a> Joshua Colp -- json: Audit ast_json_* usage for thread safety.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25585">ASTERISK-25585</a>: [patch]rasterisk never hits most of main(), but it's assumed to<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b2787876d67cb7d47ce1c7a87db515adcacc151f">[b2787876d6]</a> Walter Doekes -- main: Slight refactor of main. Improve color situation.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25552">ASTERISK-25552</a>: hashtab: Improve NULL tolerance<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afd9a89e5a0ed041d576afa1f387000404ed3c4d">[afd9a89e5a]</a> Joshua Colp -- hashtab: Add NULL check when destroying iterator.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25449">ASTERISK-25449</a>: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=07583c288828a496cd7730b55112128fea31eaef">[07583c2888]</a> Steve Davies -- Further fixes to improper usage of scheduler</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b714b2152d2ee2f6599e9decbe927d4215b6169d">[b714b2152d]</a> Matt Jordan -- res/res_rtp_asterisk: Fix assignment after ao2 decrement</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50fa9ff9972e67899dfc4e7e6766c5977d4aae7a">[50fa9ff997]</a> Matt Jordan -- Fix improper usage of scheduler exposed by 5c713fdf18f</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25546">ASTERISK-25546</a>: threadpool: Race condition between idle timeout and activation<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b818d70533916aa80af7607f225e0b1e98f41648">[b818d70533]</a> Joshua Colp -- threadpool: Handle worker thread transitioning to dead when going active.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-7803">ASTERISK-7803</a>: [patch] Update the maximum packetization values in frame.c<br/>Reported by: dea<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84ff075d411671a801f7de45d7bac48fe4f04a23">[84ff075d41]</a> Alexander Traud -- format: Update the maximum packetization time for iLBC 30.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25383">ASTERISK-25383</a>: Core dumps on startup and shutdown with MALLOC_DEBUG enabled<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028033e5a8dcf2b6d9c454786736825fe0288141">[028033e5a8]</a> Richard Mudgett -- res/ari/config.c: Fix conf_alloc() object init.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25418">ASTERISK-25418</a>: On-hold channels redirected out of a bridge appear to still be on hold<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=629458d34930e5aca56f749bc05562baf95d13f7">[629458d349]</a> Mark Michelson -- Do not swallow frames on channels leaving bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25355">ASTERISK-25355</a>: sched: ast_sched_del may return prematurely due to spurious wakeup<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85e1cb51b21c2d194647e16b81b5a1344d2ff911">[85e1cb51b2]</a> Joshua Colp -- sched: ast_sched_del may return prematurely due to spurious wakeup</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25255">ASTERISK-25255</a>: Missing AMI VarSet events when setting to an empty string.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e31cb6b2484bbf5726c59b263f13b995e60d537d">[e31cb6b248]</a> Richard Mudgett -- strings.h: Fix issues with escape string functions.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25201">ASTERISK-25201</a>: Crash in PJSIP distributor on already free'd threadpool<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=653f2087e0d2edc9df8a0154c09e2e608e13a5c5">[653f2087e0]</a> Richard Mudgett -- res_pjsip_session.c: Fix crash on call disconnect.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25212">ASTERISK-25212</a>: [patch]Segfault when using DEBUG_FD_LEAKS<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6551e16e03cc8e172d9ad8f75a040750a29a5e6e">[6551e16e03]</a> Walter Doekes -- astfd: Fix buffer overflow in DEBUG_FD_LEAKS.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22559">ASTERISK-22559</a>: gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it.<br/>Reported by: ibercom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3046bc17ed81d13899147c6e2138d0189250a8f4">[3046bc17ed]</a> ibercom -- weakref attribute detection broken with gcc 4.6 and higher</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24944">ASTERISK-24944</a>: main/audiohook.c change prevents G722 call recording<br/>Reported by: Ronald Raikes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1e8c0b9eb7807f8f0368e06e99f31937a04df70">[b1e8c0b9eb]</a> Kevin Harwell -- audiohook.c: Difference in read/write rates caused continuous buffer resets</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25083">ASTERISK-25083</a>: Message.c: Message channel becomes saturated with frames leading to spammy log messages<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=02c513058905dae19f28393ea840a47ae4a9e66d">[02c5130589]</a> Jonathan Rose -- Message.c: Clear message channel frames on cleanup</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24896">ASTERISK-24896</a>: [patch] Using force black background leads to colours not being reset<br/>Reported by: dant<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92120247e99865d97a85461b08ba763e961e6232">[92120247e9]</a> D Tucny -- term: send proper reset sequence when black background is forced</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24997">ASTERISK-24997</a>: Astobj2: Some callers of __adjust_lock do not pre-check the object<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73efb093b8cd1a81cee5f9fb0120ac7f74a82ea4">[73efb093b8]</a> Corey Farrell -- Astobj2: Ensure all calls to __adjust_lock pass a valid object.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24155">ASTERISK-24155</a>: [patch]Non-portable and non-reliable recursion detection in ast_malloc<br/>Reported by: Timo Teräs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d01706ce1ee518118456d5673f529204bdac73bb">[d01706ce1e]</a> Corey Farrell -- Improved and portable ast_log recursion avoidance</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24881">ASTERISK-24881</a>: ast_register_atexit should only be used when absolutely needed<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6adf26f14dbe2ed0addd5d07910c3b5546005987">[6adf26f14d]</a> Corey Farrell -- Replace most uses of ast_register_atexit with ast_register_cleanup.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24879">ASTERISK-24879</a>: [patch]Compilation fails due to 64bit time under OpenBSD<br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3fe43fbdc89aa51e266360dc93ed4a4445bebdb">[a3fe43fbdc]</a> snuffy -- Fix compilations errors on 64-bit OpenBSD systems</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24739">ASTERISK-24739</a>: [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules<br/>Reported by: Ed Hynan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92178247eefcf9b84bbc22a836345b43ab8770b2">[92178247ee]</a> Ed Hynan -- localtime: Fix file descriptor leak on kqueue(2) systems</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24796">ASTERISK-24796</a>: Codecs and bucket schema's prevent module unload<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=93c9c3af2f63f7f25dc24220066f2e667761e78f">[93c9c3af2f]</a> Corey Farrell -- Allow shutdown to unload modules that register bucket scheme's or codec's.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24814">ASTERISK-24814</a>: asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=54a699fb64dfb75a921146c40e507c917457b3b6">[54a699fb64]</a> Corey Farrell -- asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24740">ASTERISK-24740</a>: [patch]Segmentation fault on aoc-e event<br/>Reported by: Panos Gkikakis<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40547e7210d5ca2ba651bbcdb7af50eb330a5ed8">[40547e7210]</a> Richard Mudgett -- ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24752">ASTERISK-24752</a>: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=feddab7944a12746bcbdb6fc82cc9d4951d61eb5">[feddab7944]</a> Richard Mudgett -- HTTP: Stop accepting requests on final system shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24479">ASTERISK-24479</a>: Enable REF_DEBUG for module references<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2531f750576542037e93f4c4e087f9dbced1897d">[2531f75057]</a> Corey Farrell -- Enable REF_DEBUG for ast_module_ref / ast_module_unref.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24736">ASTERISK-24736</a>: Memory Leak Fixes<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43dd42d8aeee1eff844f605544901016047348e2">[43dd42d8ae]</a> Mark Michelson -- Fix some memory leaks.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24619">ASTERISK-24619</a>: [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ae57e0dd6841dad6930a75d7907d32a1efff4d2">[9ae57e0dd6]</a> Walter Doekes -- Fix printf problems with high ascii characters after r413586 (1.8).</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24614">ASTERISK-24614</a>: Deadlock when DEBUG_THREADS compiler flag enabled<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f12ded8871a68f3e6bbbcf7665c916c01295912">[8f12ded887]</a> Richard Mudgett -- DEBUG_THREADS: Fix regression and lock tracking initialization problems.</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24724">ASTERISK-24724</a>: 'httpstatus' Web Page Produces Incomplete HTML<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bf9d416536339e8969fe7694140a9aa6942b66f9">[bf9d416536]</a> Joshua Colp -- http: Add missing html tag to 'httpstatus' functionality.</li>
</ul><br><h4>Category: Core/Logging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25305">ASTERISK-25305</a>: Dynamic logger channels can be added multiple times<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f050fa76eb8535a2b8a3b047527a42ea369d8792">[f050fa76eb]</a> Mark Michelson -- logger: Prevent duplicate dynamic channels from being added.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25407">ASTERISK-25407</a>: Asterisk fails to log to multiple syslog destinations<br/>Reported by: Elazar Broad<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec514ad64dbc0014525008977c8c74c2856c9d3a">[ec514ad64d]</a> Elazar Broad -- core/logging: Fix logging to more than one syslog channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25112">ASTERISK-25112</a>: Logger: Configuration settings are not reset to default during reload.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9b6e228419d821a5384e3b6b056c8d5e4f30b4f4">[9b6e228419]</a> Corey Farrell -- Logger: Reset defaults before processing config.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24817">ASTERISK-24817</a>: init_logger_chain: unreachable code block<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c84dca2d82f9a3461dae3cac8e596309770586f">[4c84dca2d8]</a> Corey Farrell -- logger: Apply default console logging when configuration cannot be loaded.</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25680">ASTERISK-25680</a>: manager: manager_channelvars is not cleaned at shutdown<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f34dd104959869420cbabd2c0a4078db27e2ad22">[f34dd10495]</a> Corey Farrell -- manager: Cleanup manager_channelvars during shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25624">ASTERISK-25624</a>: AMI Event OriginateResponse bug<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe8011cc50bffd6d282a1a71e970f894ed869f5e">[fe8011cc50]</a> sungtae kim -- AMI: Fixed OriginateResponse message</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25391">ASTERISK-25391</a>: AMI GetConfigJSON returns invalid JSON<br/>Reported by: Bojan Nemčić<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74635b56381c5facaf5b5a6d12a5aa39abf10a0e">[74635b5638]</a> Ivan Poddubny -- manager: Fix GetConfigJSON returning invalid JSON</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24934">ASTERISK-24934</a>: [patch]Asterisk manager output does not escape control characters<br/>Reported by: warren smith<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e31cb6b2484bbf5726c59b263f13b995e60d537d">[e31cb6b248]</a> Richard Mudgett -- strings.h: Fix issues with escape string functions.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5d5aa67dcdc274770c47b1a801a449fb83c2f79">[f5d5aa67dc]</a> Kevin Harwell -- AMI: Escape string values.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24900">ASTERISK-24900</a>: Manager event ParkedCallSwap is not documented<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=723a9d4225d78391e91d9554b5df65424eed5969">[723a9d4225]</a> Mark Michelson -- Parking: Add documentation for AMI ParkedCallSwap event.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22670">ASTERISK-22670</a>: Asterisk crashes when processing ISDN AoC Events<br/>Reported by: klaus3000<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40547e7210d5ca2ba651bbcdb7af50eb330a5ed8">[40547e7210]</a> Richard Mudgett -- ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24721">ASTERISK-24721</a>: manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a620b287bdcbe8b4303d195fc1663aa89bca235e">[a620b287bd]</a> Jonathan Rose -- Manager: Fix Manager Action ModuleLoad to give correct response when reloading</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24049">ASTERISK-24049</a>: Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=368ecf13bf81d00b5b50d71a849116addf59a3bb">[368ecf13bf]</a> Richard Mudgett -- AMI: Revert non-backwards compatible changes from earlier commit.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b363688d4fb16f2f5cfdcffb43df28a84f1d99b">[4b363688d4]</a> Richard Mudgett -- AMI: Make AMI actions that generate event lists consistent.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1582929f9c27330ac58420e2329421a4713b70c">[c1582929f9]</a> Mark Michelson -- Prevent possible race condition on dual redirect of channels in the same bridge.</li>
</ul><br><h4>Category: Core/ManagerInterface/NewFeature</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25624">ASTERISK-25624</a>: AMI Event OriginateResponse bug<br/>Reported by: sungtae kim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe8011cc50bffd6d282a1a71e970f894ed869f5e">[fe8011cc50]</a> sungtae kim -- AMI: Fixed OriginateResponse message</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25189">ASTERISK-25189</a>: AMI: Add Linkedid header to standard channel snapshot information.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=890c92378649b99cc5281494914ec719d2bf0284">[890c923786]</a> Richard Mudgett -- AMI: Add Linkedid to the standard channel snapshot AMI event headers.</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25394">ASTERISK-25394</a>: pbx: Incorrect device and presence state when changing hint details<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1600ebca7d9a233f39ae56e8d662ba8d95a067c1">[1600ebca7d]</a> Kevin Harwell -- pbx: Deadlock between contexts container and context_merge locks</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2bd27d12223fe33b58c453965ed5c6ed3af7c4f5">[2bd27d1222]</a> Joshua Colp -- pbx: Update device and presence state when changing a hint extension.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25367">ASTERISK-25367</a>: pbx: Long pattern match hints may cause "core show hints" to crash<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc1363209e7a5b6c67bf96c593e3beb0884c1fb0">[cc1363209e]</a> Joshua Colp -- pbx: Fix crash when issuing "core show hints" with long pattern match.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25362">ASTERISK-25362</a>: Deadlock due to presence state callback<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03fe79f29eae42be24589b323a5ef3fa9259158d">[03fe79f29e]</a> Mark Michelson -- Fix deadlock on presence state changes.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25094">ASTERISK-25094</a>: PBX core: Investigate thread safety issues<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=55c8daf88b94f74e334ab246c51bdb7b469eedc4">[55c8daf88b]</a> Corey Farrell -- Fix unsafe uses of ast_context pointers.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24442">ASTERISK-24442</a>: Outgoing call files don't work properly when set in the future<br/>Reported by: tootai<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5864a358c0adf813ed0122eb32006b43fe688de">[d5864a358c]</a> Ivan Poddubny -- pbx/pbx_spool: Fix issue when call files were executed too early</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24774">ASTERISK-24774</a>: Segfault in ast_context_destroy with extensions.ael and extensions.conf<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0c82a173a4dfc78b0f5ebc484a8b905877c95fc">[f0c82a173a]</a> Matt Jordan -- main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24914">ASTERISK-24914</a>: Division by zero in file.c when playback of voicemail with video as h264<br/>Reported by: Marcello Ceschia<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2443b403417ce8eb2bb9c881120dce4214a5b2b7">[2443b40341]</a> Mark Michelson -- Ensure that a non-zero sample rate is returned for all formats.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24683">ASTERISK-24683</a>: Crash in PBX ast_hashtab_lookup_internal during core restart now<br/>Reported by: Peter Katzmann<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6adf26f14dbe2ed0addd5d07910c3b5546005987">[6adf26f14d]</a> Corey Farrell -- Replace most uses of ast_register_atexit with ast_register_cleanup.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24805">ASTERISK-24805</a>: [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6adf26f14dbe2ed0addd5d07910c3b5546005987">[6adf26f14d]</a> Corey Farrell -- Replace most uses of ast_register_atexit with ast_register_cleanup.</li>
</ul><br><h4>Category: Core/Portability</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24155">ASTERISK-24155</a>: [patch]Non-portable and non-reliable recursion detection in ast_malloc<br/>Reported by: Timo Teräs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d01706ce1ee518118456d5673f529204bdac73bb">[d01706ce1e]</a> Corey Farrell -- Improved and portable ast_log recursion avoidance</li>
</ul><br><h4>Category: Core/RTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25296">ASTERISK-25296</a>: RTP performance issue with several channel drivers.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aeeb170fc4fe4e681a96b87f8e81ade717aa2426">[aeeb170fc4]</a> Richard Mudgett -- rtp_engine.c: Fix performance issue with several channel drivers that use RTP.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84262749d2a1f59f669e801d5f796b016b223960">[84262749d2]</a> Richard Mudgett -- res_rtp_asterisk.c: Fix off-nominal crash potential.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25219">ASTERISK-25219</a>: [patch]Source and destination overlap in memcpy in rtp_engine.c<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b835312b4c2e1fcd3f7efb5ae45a0c4d10d4d5f7">[b835312b4c]</a> Walter Doekes -- rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25022">ASTERISK-25022</a>: Memory leak setting up DTLS/SRTP calls<br/>Reported by: Steve Davies<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4e207e27ef4c5db85c88969b2aec999e1897ae4">[d4e207e27e]</a> Matt Jordan -- main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b6410c4f880d8f715621de2c95b07ea3489d853">[0b6410c4f8]</a> Steve Davies -- res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25811">ASTERISK-25811</a>: Unable to delete object from sorcery cache<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=48d713a83232fc51fc95d2e50a72c67980e63354">[48d713a832]</a> gtjoseph -- sorcery: Refactor create, update and delete to better deal with caches</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32fc784284b570a05841d95c6d9a373b4bf3a35d">[32fc784284]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25625">ASTERISK-25625</a>: res_sorcery_memory_cache: Add full backend caching<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59d5bb0613810418f2a618b9a6dee5bcfd45767e">[59d5bb0613]</a> Joshua Colp -- res_sorcery_memory_cache: Add support for a full backend cache.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25165">ASTERISK-25165</a>: Testsuite - Sorcery memory cache leaks<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc45f4040df77fda402a50822f027f870d114913">[fc45f4040d]</a> Richard Mudgett -- res_sorcery_realtime.c: Fix crash from NULL sorcery object type.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=156395e743189649280066c1497292bb97ed022d">[156395e743]</a> Mark Michelson -- res_sorcery_realtime: Fix leak of sorcery object type.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24996">ASTERISK-24996</a>: chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf<br/>Reported by: Ashley Sanders<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3278fe5327c6d34966ef6c87191d8d7d4fae4f66">[3278fe5327]</a> Ashley Sanders -- chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24612">ASTERISK-24612</a>: res_pjsip: No information if a required sorcery wizard is not loaded<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=758a897876ee5c8264d6a2b1ac03aff16215ae3a">[758a897876]</a> Joshua Colp -- sorcery: Output an error message if a wizard is specified for an object type and it isn't found.</li>
</ul><br><h4>Category: Core/Stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25137">ASTERISK-25137</a>: endpoint stasis messages are delivered twice<br/>Reported by: Vitezslav Novy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4a566918a4bcf01269f321f0979a7bf4c5c33d6">[e4a566918a]</a> Matt Jordan -- tests/test_stasis_endpoints: Remove expected duplicate events</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a1c4885be4eeb3f6c631b0cfb4709442c9464ae">[3a1c4885be]</a> gtjoseph -- endpoint/stasis: Eliminate duplicate events on endpoint status change</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=35c699086ae2fd81b2473307ccb2ae79ad32375a">[35c699086a]</a> gtjoseph -- endpoint/stasis: Eliminate duplicate events on endpoint status change</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25121">ASTERISK-25121</a>: Stasis: Fix unsafe use of stasis_unsubscribe in modules.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d266cbe025e30ce18121f43dbb6b11620b4d5e1">[0d266cbe02]</a> Corey Farrell -- Stasis: Fix unsafe use of stasis_unsubscribe in modules.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24682">ASTERISK-24682</a>: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=894d4d781cd361a87ffc0ed343b47345bbfeaaaf">[894d4d781c]</a> Matt Jordan -- apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values</li>
</ul><br><h4>Category: Core/UDPTL</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25603">ASTERISK-25603</a>: [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c6b1b2b1c84b2b9d27349036f2760aa263d74ef5">[c6b1b2b1c8]</a> Richard Mudgett -- AST-2016-003 udptl.c: Fix uninitialized values.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25742">ASTERISK-25742</a>: Secondary IFP Packets can result in accessing uninitialized pointers and a crash<br/>Reported by: Torrey Searle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c6b1b2b1c84b2b9d27349036f2760aa263d74ef5">[c6b1b2b1c8]</a> Richard Mudgett -- AST-2016-003 udptl.c: Fix uninitialized values.</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24097">ASTERISK-24097</a>: Documentation - CHANNEL function help text missing 'linkedid' argument<br/>Reported by: Steven T. Wheeler<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=018ccf680b231c5d7124faeeac6ce0bfe6d3da61">[018ccf680b]</a> Rusty Newton -- func_channel: Add help text for undocumented CHANNEL function arguments</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25373">ASTERISK-25373</a>: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d1bdb9d3b1c993b98fdf5041c11708742867820">[6d1bdb9d3b]</a> Walter Doekes -- func_callerid: Document that CALLERID(pres) is available.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25527">ASTERISK-25527</a>: Quirky xmldoc description wrapping<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d425f2eb439161065b43d1178bfdac632d8bf56">[0d425f2eb4]</a> Walter Doekes -- xmldoc: Improve xmldoc wrapping of 'core show ...' output.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24867">ASTERISK-24867</a>: Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62c64c3bd1938280cf671afd6708d6026c0b8e49">[62c64c3bd1]</a> Rusty Newton -- Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24853">ASTERISK-24853</a>: Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true)<br/>Reported by: PSDK<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=62c64c3bd1938280cf671afd6708d6026c0b8e49">[62c64c3bd1]</a> Rusty Newton -- Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24085">ASTERISK-24085</a>: Documentation - We should remove or further document the 'contact' section in pjsip.conf<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7890d0ad0718ad25417d765f6fb4f4f76582006d">[7890d0ad07]</a> Joshua Colp -- pjsip: Remove "contact" type from pjsip.conf.sample</li>
</ul><br><h4>Category: Features</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25003">ASTERISK-25003</a>: Asterisk crashes on attended transfer (using feature)<br/>Reported by: Artem Volodin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be1260a35f88faea4fa029d59343b124d250a8a6">[be1260a35f]</a> Richard Mudgett -- features: Fix crash when transferee hangs up during DTMF attended transfer.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23841">ASTERISK-23841</a>: DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a356232bdf2cd231766459f476f5f0885981878">[7a356232bd]</a> Richard Mudgett -- DTMF atxfer: Setup recall channels as if the transferee initiated the call.</li>
</ul><br><h4>Category: Formats/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25664">ASTERISK-25664</a>: ast_format_cap_append_by_type leaks a reference<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=52e9de001688025669eff569be69b4c9c9191bf5">[52e9de0016]</a> Corey Farrell -- ast_format_cap_append_by_type: Resolve codec reference leak.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25584">ASTERISK-25584</a>: [patch] format-attribute module: VP8 missing<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a18193dc09e90102f664a94fc9eeea5cac44b71">[5a18193dc0]</a> Alexander Traud -- res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25545">ASTERISK-25545</a>: [patch] translation module gets cached not joint format<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b508789ab225f57e22bce93e243e9d642a73191">[0b508789ab]</a> Alexander Traud -- translate: Provide translation modules the result of SDP negotiation.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25535">ASTERISK-25535</a>: [patch] format creation on module load instead of cache<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e5bf12b33de8db9c53f571e4c4d5cb094a0d008">[4e5bf12b33]</a> Joshua Colp -- format_cap: Don't append the 'none' format when appending all.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f3ac4d8090207dd4440bf279e1d5ce4702aee314">[f3ac4d8090]</a> Alexander Traud -- ast_format_cap: Avoid format creation on module load, use cache instead.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25537">ASTERISK-25537</a>: [patch] format-attribute module: RFC or internal defaults?<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4bf84459c77227b7adc642d04b9ad93659d96ee2">[4bf84459c7]</a> Alexander Traud -- rtp_engine: Init a format-attribute module to its RFC defaults.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25533">ASTERISK-25533</a>: [patch] buffer for ast_format_cap_get_names only 64 bytes<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bff400df7ff1fda353fab49de2fcf9cbba5cd89">[1bff400df7]</a> Alexander Traud -- ast_format_cap_get_names: To display all formats, the buffer was increased.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25054">ASTERISK-25054</a>: Formats interface's cannot be unregistered, needs to hold modules until shutdown.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f38066fcad559da3f214b4d309dcc4b070665d66">[f38066fcad]</a> Corey Farrell -- Format Interfaces: Prevent unload except by shutdown.</li>
</ul><br><h4>Category: Formats/format_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25573">ASTERISK-25573</a>: [patch] H.264 format attribute module: resets whole SDP<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1aa552b2a28d51ef9d6ac4f236ee9852b0ca449a">[1aa552b2a2]</a> Alexander Traud -- res_format_attr_h264: Do not reset string buffer.</li>
</ul><br><h4>Category: Functions/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17608">ASTERISK-17608</a>: func_aes.so cannot be loaded if res_crypto / openssl not compiled<br/>Reported by: Warren Selby<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0e4b997cd7b1efe42aec900778efee46b7040365">[0e4b997cd7]</a> Corey Farrell -- res_monitor: Add dependency on func_periodic_hook.</li>
</ul><br><h4>Category: Functions/func_callerid</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25373">ASTERISK-25373</a>: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants<br/>Reported by: Walter Doekes<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d1bdb9d3b1c993b98fdf5041c11708742867820">[6d1bdb9d3b]</a> Walter Doekes -- func_callerid: Document that CALLERID(pres) is available.</li>
</ul><br><h4>Category: Functions/func_cdr</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25179">ASTERISK-25179</a>: CDR(billsec,f) and CDR(duration,f) report incorrect values<br/>Reported by: Gianluca Merlo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46b2de55f95927b67f9309426353f2f5f5d7c830">[46b2de55f9]</a> Matt Jordan -- funcs/func_cdr: Correctly report high precision values for duration and billsec</li>
</ul><br><h4>Category: Functions/func_channel</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24097">ASTERISK-24097</a>: Documentation - CHANNEL function help text missing 'linkedid' argument<br/>Reported by: Steven T. Wheeler<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=018ccf680b231c5d7124faeeac6ce0bfe6d3da61">[018ccf680b]</a> Rusty Newton -- func_channel: Add help text for undocumented CHANNEL function arguments</li>
</ul><br><h4>Category: Functions/func_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-18708">ASTERISK-18708</a>: func_curl hangs channel under load<br/>Reported by: Dave Cabot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7c6bedb061b6e609d8ec0a1a9c67e1ba6ba194d">[f7c6bedb06]</a> Joshua Colp -- func_curl: Don't hold exclusive lock when performing HTTP request.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24676">ASTERISK-24676</a>: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25a67d561c521d057312454965bbffe9074703cf">[25a67d561c]</a> Mark Michelson -- Multiple revisions 431297-431298</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24672">ASTERISK-24672</a>: [PATCH] Memory leak in func_curl CURLOPT<br/>Reported by: Kristian Høgh<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc993db55c7bb91c5470c0fb54009c95faae357b">[dc993db55c]</a> Kristian Hogh -- funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyed</li>
</ul><br><h4>Category: Functions/func_dialplan</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-21765">ASTERISK-21765</a>: [patch] - FILE function's length argument counts from beginning of file rather than the offset<br/>Reported by: John Zhong<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73dcea59bd17c694e636a886e151863c02311822">[73dcea59bd]</a> Matt Jordan -- funcs/func_env: Fix regression caused in FILE read operation</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=37d33ed997132e97e45f7838da84ec5ce6fae9c3">[37d33ed997]</a> Di-Shi Sun -- FILE: fix retrieval of file contents when offset is specified</li>
</ul><br><h4>Category: Functions/func_iconv</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25272">ASTERISK-25272</a>: [patch]The ICONV dialplan function sometimes returns garbage<br/>Reported by: Etienne Lessard<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=962a9d61f8666607b25fd3307cd938576ee60da0">[962a9d61f8]</a> Etienne Lessard -- func_iconv: Ensure output strings are properly terminated.</li>
</ul><br><h4>Category: Functions/func_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22708">ASTERISK-22708</a>: res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work<br/>Reported by: JoshE<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3dcec04ab5b35d3085bd2a3ca2c5af575ed3951b">[3dcec04ab5]</a> Martin Tomec -- res_odbc: Use negative connection cache for all connections</li>
</ul><br><h4>Category: Functions/func_periodic_hook</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25085">ASTERISK-25085</a>: [patch]Potential crash after unload of func_periodic_hook or test_message<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b7282ca4037114ac19da5149dce958f837d1422">[6b7282ca40]</a> Corey Farrell -- Fix potential crash after unload of func_periodic_hook or test_message.</li>
</ul><br><h4>Category: Functions/func_talkdetect</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24988">ASTERISK-24988</a>: func_talkdetect: Test is bouncing sporadically<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5ac65ddfb46f715490b4eccbef57dc6f424e9bc2">[5ac65ddfb4]</a> Matt Jordan -- res/ari: Register Stasis application on WebSocket attempt</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31cc24aad603716e7ef68b384faebd14e0dfa460">[31cc24aad6]</a> Matt Jordan -- res/res_http_websocket: Add a pre-session established callback</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23666">ASTERISK-23666</a>: CLONE - nested functions aren't portable<br/>Reported by: Diederik de Groot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5bc032567ac1af93e8d5320a7a6c126a8639611">[f5bc032567]</a> Diederik de Groot -- Add support for the clang compiler; update RAII_VAR to use BlocksRuntime</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24663">ASTERISK-24663</a>: [patch] Unnamed semaphore autoconf check fails on cross compilation<br/>Reported by: abelbeck<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f6630e248124bf34683d20f829d63a9b6dfc8c79">[f6630e2481]</a> abelbeck -- configure: If cross-compiling, assume we have working semaphores</li>
</ul><br><h4>Category: PBX/pbx_config</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25061">ASTERISK-25061</a>: pbx_config: Register manager actions with module version of macro.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=69ae8cf0a4c441c1ebfab0da423cc1acfcc8c7aa">[69ae8cf0a4]</a> Corey Farrell -- pbx_config: Register manager actions with module version of macro.</li>
</ul><br><h4>Category: PBX/pbx_dundi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25677">ASTERISK-25677</a>: pbx_dundi: leaks during failed load.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=79b430988102eefed7d2ddb3655361df129a6216">[79b4309881]</a> Corey Farrell -- pbx_dundi: Run cleanup on failed load.</li>
</ul><br><h4>Category: Resources/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25584">ASTERISK-25584</a>: [patch] format-attribute module: VP8 missing<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a18193dc09e90102f664a94fc9eeea5cac44b71">[5a18193dc0]</a> Alexander Traud -- res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25441">ASTERISK-25441</a>: Deadlock in res_sorcery_memory_cache.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40c69e78f571781b67a240554c119b870e3cd6d4">[40c69e78f5]</a> Richard Mudgett -- res_sorcery_memory_cache.c: Fix deadlock with scheduler.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dfeb513e85d13550d81b40df5e95333c1ad5c61c">[dfeb513e85]</a> Richard Mudgett -- res_sorcery_memory_cache.c: Replace inline code with function.</li>
</ul><br><h4>Category: Resources/res_agi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25593">ASTERISK-25593</a>: fastagi: record file closed after sending result<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=45efbf8503a29d298a9cb6c5de4925037a642b35">[45efbf8503]</a> Kevin Harwell -- fastagi: record file closed after sending result</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23390">ASTERISK-23390</a>: NewExten Event with application AGI shows up before and after AGI runs<br/>Reported by: Benjamin Keith Ford<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=055001716cd6548ec644f3cc62841b3b7ff4f66c">[055001716c]</a> Graham Barnett -- app_voicemail: Fix crash with IMAP backends when greetings aren't present</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25882">ASTERISK-25882</a>: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b6c4decd31377f5d250a41e52d300bd5d1b092d">[7b6c4decd3]</a> Richard Mudgett -- res_stasis: Fix crash on a hanging up channel.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25771">ASTERISK-25771</a>: ARI:Crash - Attended transfers of channels into Stasis application.<br/>Reported by: Javier Riveros <ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=40d9e9e2384fcf1da715d7a34680f02631c8ec64">[40d9e9e238]</a> Kevin Harwell -- bridge.c: Crash during attended transfer when missing a local channel half</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ff3da61c35f44761516e29f315cefd908ad895d8">[ff3da61c35]</a> Kevin Harwell -- res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25683">ASTERISK-25683</a>: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG <br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=935d641f3bb71d786e69ff99a6b6b99165fd4ca2">[935d641f3b]</a> Mark Michelson -- Remove res/ari/* content during 'make clean'.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25522">ASTERISK-25522</a>: ARI: Crash when creating channel via ARI originate with requesting channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=506aea26e6c67cd53874aa3ffef278524dfd7878">[506aea26e6]</a> Matt Jordan -- main/dial: Protect access to the format_cap structure of the requesting channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25325">ASTERISK-25325</a>: ARI PUT reload chan_sip HTTP response 404<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=865377fc38134234f17def6634c47a989cf0e77a">[865377fc38]</a> Rodrigo Ramírez Norambuena -- chan_sip.c: Validation on module reload</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25091">ASTERISK-25091</a>: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge<br/>Reported by: Ilya Trikoz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9015bb4c8c9882de35066c6586189ab78268a12f">[9015bb4c8c]</a> Mark Michelson -- Resolve race conditions involving Stasis bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24812">ASTERISK-24812</a>: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d1a1533bf8a12c33419695227884cc8d386b943">[3d1a1533bf]</a> Matt Jordan -- ARI/PJSIP: Apply requesting channel's format cap to created channels</li>
</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25091">ASTERISK-25091</a>: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge<br/>Reported by: Ilya Trikoz<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9015bb4c8c9882de35066c6586189ab78268a12f">[9015bb4c8c]</a> Mark Michelson -- Resolve race conditions involving Stasis bridges.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24560">ASTERISK-24560</a>: Creating a named ARI bridge twice causes a crash<br/>Reported by: Kinsey Moore<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a7ba8a58a86235eaf88850df03b6865413525b4e">[a7ba8a58a8]</a> Ashley Sanders -- ARI: Fixed crash that occurred when updating a bridge when the optional query parameter 'name' was not supplied.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24591">ASTERISK-24591</a>: Stasis() side of an ARI originated channel cannot be Redirected<br/>Reported by: Kinsey Moore<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f3c60cee79400684d073c159fb29701e2cb11cb">[8f3c60cee7]</a> Kinsey Moore -- ARI: Allow usage of ASYNCGOTO with Stasis()</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25522">ASTERISK-25522</a>: ARI: Crash when creating channel via ARI originate with requesting channel<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=506aea26e6c67cd53874aa3ffef278524dfd7878">[506aea26e6]</a> Matt Jordan -- main/dial: Protect access to the format_cap structure of the requesting channel</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24812">ASTERISK-24812</a>: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d1a1533bf8a12c33419695227884cc8d386b943">[3d1a1533bf]</a> Matt Jordan -- ARI/PJSIP: Apply requesting channel's format cap to created channels</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24677">ASTERISK-24677</a>: ARI GET variable on channel provides unhelpful response on non-existent variable<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f726304283251f811b4d73e6d958baac45e37efa">[f726304283]</a> Joshua Colp -- res_ari_channels: Return a 404 response when a requested channel variable does not exist.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24591">ASTERISK-24591</a>: Stasis() side of an ARI originated channel cannot be Redirected<br/>Reported by: Kinsey Moore<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f3c60cee79400684d073c159fb29701e2cb11cb">[8f3c60cee7]</a> Kinsey Moore -- ARI: Allow usage of ASYNCGOTO with Stasis()</li>
</ul><br><h4>Category: Resources/res_calendar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25679">ASTERISK-25679</a>: res_calendar leaks scheduler.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d3a1167fca6b6e9d55f355b485ff4ad19ba23e8">[1d3a1167fc]</a> Corey Farrell -- res_calendar: Cleanup scheduler context at unload.</li>
</ul><br><h4>Category: Resources/res_config_curl</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24676">ASTERISK-24676</a>: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=25a67d561c521d057312454965bbffe9074703cf">[25a67d561c]</a> Mark Michelson -- Multiple revisions 431297-431298</li>
</ul><br><h4>Category: Resources/res_config_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24808">ASTERISK-24808</a>: res_config_odbc: Improper escaping of backslashes occurs with MySQL<br/>Reported by: Javier Acosta<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afea98dc735a6a7a76b21edc93174f1102329165">[afea98dc73]</a> Javier Acosta -- res/res_config_odbc: Fix improper escaping of backslashes with MySQL</li>
</ul><br><h4>Category: Resources/res_config_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><br><h4>Category: Resources/res_corosync</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24998">ASTERISK-24998</a>: res_corosync: res_corosync tries to load even if res_corosync.conf is missing<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1474bb05f64c2b1db4417667a4d27576dbe219a7">[1474bb05f6]</a> gtjoseph -- res_corosync: Add check for config file before calling corosync apis</li>
</ul><br><h4>Category: Resources/res_crypto</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25673">ASTERISK-25673</a>: res_crypto leaks CLI entries<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5406b1f9e5fdad87a91c82d6ee39249472bd0dc">[a5406b1f9e]</a> Corey Farrell -- res_crypto: Perform cleanup at shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24550">ASTERISK-24550</a>: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake<br/>Reported by: Osaulenko Alexander<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><br><h4>Category: Resources/res_fax</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22790">ASTERISK-22790</a>: check_modem_rate() may return incorrect rate for V.27<br/>Reported by: not here<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3fb6daeb551bf4fec13820a2d2f9df7faf287b9b">[3fb6daeb55]</a> Kevin Harwell -- res_fax: allow 2400 transmission rate according to v.27ter standard</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23231">ASTERISK-23231</a>: Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load<br/>Reported by: David Brillert<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3fb6daeb551bf4fec13820a2d2f9df7faf287b9b">[3fb6daeb55]</a> Kevin Harwell -- res_fax: allow 2400 transmission rate according to v.27ter standard</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24955">ASTERISK-24955</a>: res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3fb6daeb551bf4fec13820a2d2f9df7faf287b9b">[3fb6daeb55]</a> Kevin Harwell -- res_fax: allow 2400 transmission rate according to v.27ter standard</li>
</ul><br><h4>Category: Resources/res_format_attr_h264</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24616">ASTERISK-24616</a>: Crash in res_format_attr_h264 due to invalid string copy<br/>Reported by: Yura Kocyuba<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f67402a52ad2b9d17754252d8de82902c1bfc760">[f67402a52a]</a> Joshua Colp -- res_format_attr_h264: Fix crash when determining joint capability.</li>
</ul><br><h4>Category: Resources/res_format_attr_opus</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25583">ASTERISK-25583</a>: [patch] format-attribute module: RFC 7587 (Opus Codec)<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3e2178c05e5e324482f1fb46e488e96e574284cd">[3e2178c05e]</a> Alexander Traud -- res_format_attr_opus: Update to latest RFC 7587.</li>
</ul><br><h4>Category: Resources/res_http_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24972">ASTERISK-24972</a>: Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server<br/>Reported by: Alex A. Welzl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f8acadde2c1fcd36fa56bb39bf6058d7116f0962">[f8acadde2c]</a> Joshua Colp -- AST-2016-001 http: Provide greater control of TLS and set modern defaults.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24106">ASTERISK-24106</a>: WebSockets Automatically decides what driver it will use <br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0393bd6bed3b87b312d7fc252c4fa3782df8260a">[0393bd6bed]</a> Corey Farrell -- chan_sip: Allow websockets to be disabled.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25312">ASTERISK-25312</a>: res_http_websocket: Terminate connection on fatal cases<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4e9416138339274176cb87b26db905723e553ba">[b4e9416138]</a> Joshua Colp -- res_http_websocket: Forcefully terminate on write errors.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24963">ASTERISK-24963</a>: ASAN: heap-use-after-free with PJSIP and WSS<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8af6c9cf6bc9f8217fd96f59a6f248330583bdb9">[8af6c9cf6b]</a> Ivan Poddubny -- res_pjsip_transport_websocket: Fix use-after-free bugs.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24566">ASTERISK-24566</a>: Uninit buf in WS write<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b054bdc6dfce762cdb26608bd7a7b90484150ba">[4b054bdc6d]</a> Richard Mudgett -- res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write().</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24472">ASTERISK-24472</a>: Asterisk Crash in OpenSSL when calling over WSS from JSSIP<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb768ec33a27cc9de76ee7e95ef9f08e0948b47b">[fb768ec33a]</a> Joshua Colp -- res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero.</li>
</ul><br><h4>Category: Resources/res_jabber</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-14233">ASTERISK-14233</a>: [patch] Buddies are always auto-registered when processing the roster<br/>Reported by: Simon Arlott<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05de9082a599df079ab1f1eb307b60d04a5c3e3b">[05de9082a5]</a> Simon Arlott -- res_xmpp: Buddies are always auto-registered when processing the roster</li>
</ul><br><h4>Category: Resources/res_musiconhold</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25687">ASTERISK-25687</a>: res_musiconhold: Concurrent invocations of 'moh reload' cause a crash<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7cfda0b384897c6f52db0c7d7b126f5158461cb">[e7cfda0b38]</a> Sean Bright -- res_musiconhold: Prevent multiple simultaneous reloads.</li>
</ul><br><h4>Category: Resources/res_mwi_external_ami</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25117">ASTERISK-25117</a>: res_mwi_external_ami: Fix manager action registrations.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7edb59db6dfb543300f43c8055adda4ab1fd1c9">[e7edb59db6]</a> Corey Farrell -- res_mwi_external_ami: Use module version of AMI registration.</li>
</ul><br><h4>Category: Resources/res_odbc</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22708">ASTERISK-22708</a>: res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work<br/>Reported by: JoshE<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3dcec04ab5b35d3085bd2a3ca2c5af575ed3951b">[3dcec04ab5]</a> Martin Tomec -- res_odbc: Use negative connection cache for all connections</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24742">ASTERISK-24742</a>: [patch] Fix ast_odbc_find_table function in res_odbc<br/>Reported by: ibercom<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85824113447c8507efa116c5a1a988bcddff71f5">[8582411344]</a> ibercom -- res/res_odbc: Remove unneeded queries when determining if a table exists</li>
</ul><br><h4>Category: Resources/res_parking</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25369">ASTERISK-25369</a>: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fbf720db91ae8942e9c2ba092179ab2352d44b06">[fbf720db91]</a> Jonathan Rose -- ParkAndAnnounce: Add variable inheritance</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25254">ASTERISK-25254</a>: Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c782320c68633c9b1b805affaec1bfe604370d7f">[c782320c68]</a> Richard Mudgett -- res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24899">ASTERISK-24899</a>: Parking fall-through behavior different in 13<br/>Reported by: Malcolm Davenport<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f25076f67c345e85f16cc859cf9f06ecab0c870">[0f25076f67]</a> Mark Michelson -- ParkedCall: Don't allow dialplan fallthrough after retrieving parked call.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-23850">ASTERISK-23850</a>: Park Application does not respect Return Context Priority<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f94b9674914467f68e1239f6999bbf5e3b7f292">[1f94b96749]</a> Richard Mudgett -- app_macro: Don't restore the calling location on a channel redirect.</li>
</ul><br><h4>Category: Resources/res_phoneprov</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25721">ASTERISK-25721</a>: [patch] res_phoneprov: memory leak and heap-use-after-free<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20e9792fbc7eb8c5ca23936bac7793e2f9be37c3">[20e9792fbc]</a> Badalyan Vyacheslav -- Resources/res_phoneprov: fix memory leak and heap-use-after-free</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25829">ASTERISK-25829</a>: res_pjsip: PJSIP does not accept spaces when separating multiple AORs<br/>Reported by: Mateusz Kowalski<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=530cff5f5f3be4d2f53e84a656f10f2f43638d1c">[530cff5f5f]</a> gtjoseph -- res_pjsip: Strip spaces from items parsed from comma-separated lists</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25727">ASTERISK-25727</a>: RPM build requires OPTIONAL_API cflag due to PJSIP requirement<br/>Reported by: Gergely Dömsödi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0f306203160c0799ecaf00f539dbb31daba1d17">[c0f3062031]</a> gtjoseph -- res_statsd: Fix exports.in for missing symbols</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25337">ASTERISK-25337</a>: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub<br/>Reported by: Jacques Peacock<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34c64707d1aa346fb0e9c7f97e375d22dedf67d9">[34c64707d1]</a> gtjoseph -- res_pjsip_caller_id: Fix segfault when replacing rpid or pai header</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25751">ASTERISK-25751</a>: res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c1bf014ea08cf66835a6f000e2bd6c7da588da6b">[c1bf014ea0]</a> gtjoseph -- res_pjsip: Handle pjsip_dlg_create_uas deprecation</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25606">ASTERISK-25606</a>: Core dump when using transports in sorcery<br/>Reported by: Martin Moučka<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2451d4e4550336197ee2e482750cc53f30afa352">[2451d4e455]</a> gtjoseph -- res_pjsip: Fix infinite recursion when loading transports from realtime</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25702">ASTERISK-25702</a>: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2<br/>Reported by: Nic Colledge<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32fc784284b570a05841d95c6d9a373b4bf3a35d">[32fc784284]</a> Alexei Gradinari License #5691 -- res_sorcery_realtime: Fix regex regression.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25712">ASTERISK-25712</a>: Second call to already-on-call phone and Asterisk sends "Ready"<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=02035212de98926c95fec2ac53ccc3691aa1de8f">[02035212de]</a> Richard Mudgett -- res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25686">ASTERISK-25686</a>: PJSIP: qualify_timeout is a double, database schema is an integer<br/>Reported by: Marcelo Terres<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=46f21df302054df826c989511f4dc0f208a82beb">[46f21df302]</a> Daniel Journo -- pjsip/alembic: Fix qualify_timeout column definition</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25668">ASTERISK-25668</a>: res_pjsip: Deadlock in distributor<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=96094feab6185de6fb26ad3a6dfe908f34d6cdcf">[96094feab6]</a> Mark Michelson -- PJSIP: Prevent deadlock due to dialog/transaction lock inversion.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25116">ASTERISK-25116</a>: res_pjsip: Two PeerStatus AMI messages are sent for every status change<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a1c4885be4eeb3f6c631b0cfb4709442c9464ae">[3a1c4885be]</a> gtjoseph -- endpoint/stasis: Eliminate duplicate events on endpoint status change</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=35c699086ae2fd81b2473307ccb2ae79ad32375a">[35c699086a]</a> gtjoseph -- endpoint/stasis: Eliminate duplicate events on endpoint status change</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25608">ASTERISK-25608</a>: res_pjsip/contacts/statsd: Lifecycle events aren't consistent<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=450579e908279e664cb4364a2e7dd1cfd6a90396">[450579e908]</a> gtjoseph -- res_pjsip/contacts/statsd: Make contact lifecycle events more consistent</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25595">ASTERISK-25595</a>: Unescaped : in messge sent to statsd<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9184fbeb347d1168add1f3140af3b6837c8d78db">[9184fbeb34]</a> gtjoseph -- res_pjsip: Use a MD5 hash for static Contact IDs</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25598">ASTERISK-25598</a>: res_pjsip: Contact status messages are printing a hash instead of the uri<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed9134282e22c6985ce853f53d7569aa5b93ebe0">[ed9134282e]</a> gtjoseph -- res_pjsip: Update logging to show contact->uri in messages</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25486">ASTERISK-25486</a>: res_pjsip: Fix deadlock when validating URIs<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f2725c8b77f6e6d6b70c12c4e57e26083530c3be">[f2725c8b77]</a> Joshua Colp -- res_pjsip: Move URI validation to use time.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25455">ASTERISK-25455</a>: Deadlock of PJSIP realtime over res_config_pgsql <br/>Reported by: mdu113<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc6ec661b3f109e196a60f1285d6554f25efa12f">[dc6ec661b3]</a> mdu113 -- res_config_pgsql.c: Fix deadlock loading realtime configuration.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25295">ASTERISK-25295</a>: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5469caa9ddf002d2e75b5fe5dec0c4dbebea1d1e">[5469caa9dd]</a> Joshua Colp -- res_pjsip: Use hash for contact object identity instead of Contact URI.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a676ba2aad5525926ae31b8317b95ae52cbbabbb">[a676ba2aad]</a> Joshua Colp -- taskprocessor: Fix race condition between unreferencing and finding.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25381">ASTERISK-25381</a>: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3e6debdb95a5895894ed2b58b600fcdf17927b9">[c3e6debdb9]</a> Matt Jordan -- res/res_pjsip: Purge contacts when an AoR is deleted</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25339">ASTERISK-25339</a>: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc6fe07f5c114bdeaef4a3b83a11faaa9d1046eb">[bc6fe07f5c]</a> Matt Jordan -- res_pjsip/pjsip_configuration: Disregard empty auth values</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25304">ASTERISK-25304</a>: res_pjsip: XML sanitization may write past buffer<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8521a86367ac6090210a89878c8fee6d19c43642">[8521a86367]</a> Joshua Colp -- res_pjsip: Ensure sanitized XML is NULL terminated.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25201">ASTERISK-25201</a>: Crash in PJSIP distributor on already free'd threadpool<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=653f2087e0d2edc9df8a0154c09e2e608e13a5c5">[653f2087e0]</a> Richard Mudgett -- res_pjsip_session.c: Fix crash on call disconnect.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25168">ASTERISK-25168</a>: Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c<br/>Reported by: Carl Fortin<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0d67e04359505a06409d8211bb0c2b65fe82125f">[0d67e04359]</a> Richard Mudgett -- res_pjsip_mwi.c: Fix MWI subscription memory corruption crash.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0422433f4722e6e692b0c84342e048feff344e80">[0422433f47]</a> Richard Mudgett -- PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8ea214aed782424a884b9a2f67d6dca270854e83">[8ea214aed7]</a> Richard Mudgett -- PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25115">ASTERISK-25115</a>: Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ada7346792452f021911063668997f79fdabc1f1">[ada7346792]</a> Richard Mudgett -- res_pjsip: Need to use the same serializer for a pjproject SIP transaction.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25171">ASTERISK-25171</a>: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e4a2ef9e4ef27488609bb01fc55e965cd93a9ad5">[e4a2ef9e4e]</a> Joshua Colp -- channel: Remove ignore of answer on non-outgoing channels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25158">ASTERISK-25158</a>: res_pjsip: Add option to use AAL2 packing when negotiating g.726<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=31c77b157b84527b1a68d96f7a23c3e7b242ee99">[31c77b157b]</a> Kevin Harwell -- res_pjsip: Add option to force G.726 to be treated as AAL2 packed.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25096">ASTERISK-25096</a>: [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h)<br/>Reported by: Josh Kitchens<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8af6c9cf6bc9f8217fd96f59a6f248330583bdb9">[8af6c9cf6b]</a> Ivan Poddubny -- res_pjsip_transport_websocket: Fix use-after-free bugs.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25131">ASTERISK-25131</a>: chan_pjsip: In-dialog authentication not handled.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe21f2e52f1b6629a254cc2f34345c4de6ec4293">[fe21f2e52f]</a> Richard Mudgett -- res_pjsip_session: Fix in-dialog authentication.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25105">ASTERISK-25105</a>: res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=60e2fbfe624680d7df948aab243d77ff111e4f4e">[60e2fbfe62]</a> gtjoseph -- res_pjsip: Refactor endpt_send_transaction (qualify_timeout)</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25089">ASTERISK-25089</a>: res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd78ab42e4316c53fa517ca19956d9be38f368a5">[dd78ab42e4]</a> gtjoseph -- res_pjsip_config_wizard/config: Fix template processing</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25033">ASTERISK-25033</a>: Asterisk 13 (branch head) won't compile without PJSip<br/>Reported by: Peter Whisker<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=63196a825673a06f84df90314e48be5aa3f67d06">[63196a8256]</a> Corey Farrell -- res_pjsip_dlg_options: Fix MODULEINFO section.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25020">ASTERISK-25020</a>: Mismatched response to outgoing REGISTER request<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e39bd6ba463f25be2265228f835957424563642b">[e39bd6ba46]</a> Mark Michelson -- res_pjsip_outbound_registration: Don't fail on delayed processing: 13.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bf008fc760a178e8faeb34f033220531ddb5b08">[1bf008fc76]</a> Mark Michelson -- res_pjsip_outbound_registration: Add debugging messages.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24999">ASTERISK-24999</a>: PJSIP crashes with malformed contact line<br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f70d21b2cf15e9b84beb1f495362cd896c520a9b">[f70d21b2cf]</a> gtjoseph -- res_pjsip: Validate that contact uris start with sip: or sips:</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24977">ASTERISK-24977</a>: Contacts that don't use qualify are being marked as unavailable<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=63169e00ff21d528e40568dec6dcfd0114b55c48">[63169e00ff]</a> gtjoseph -- pjsip_options: Fix non-qualified contacts showing as unavailable</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24863">ASTERISK-24863</a>: res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=674b18bdf0923776fae692575869640f6c00e0b1">[674b18bdf0]</a> gtjoseph -- pjsip_options: Add qualify_timeout processing and eventing</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bf46799f0e9836fc6c84b9c814faef4156e625d9">[bf46799f0e]</a> gtjoseph -- res_pjsip: Refactor endpt_send_request to include transaction timeout</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b6f6ff841634fcb68b473fde160f1100f70f855">[1b6f6ff841]</a> gtjoseph -- res_pjsip: Add global option to limit the maximum time for initial qualifies</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24380">ASTERISK-24380</a>: core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a63f7ad04a00bfe8e6b47fd1a60de6691813c0d1">[a63f7ad04a]</a> Richard Mudgett -- translate.c: Only select audio codecs to determine the best translation choice.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24935">ASTERISK-24935</a>: res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75c2c859620edc9bfc03fada26bb83e90013b33e">[75c2c85962]</a> gtjoseph -- res_pjsip_phoneprov_provider: Fix reference leak on unload</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1102cd64282487dbc260672d6a9679ca059cd70">[b1102cd642]</a> Corey Farrell -- res_pjsip_phoneprov_provider: Revert 433996 / 433997.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=709fa14b44cf90cdfcb5098c90455a8cbff013c0">[709fa14b44]</a> Corey Farrell -- res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24910">ASTERISK-24910</a>: "timer=no" and "timer=required" settings in pjsip.conf fail<br/>Reported by: Ray Crumrine<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2270c40d33827a605b500e7afe01be9976b1f5e0">[2270c40d33]</a> Kevin Harwell -- res_pjsip: config option 'timers' can't be set to 'no'</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24920">ASTERISK-24920</a>: Asterisk handles duplicate SIP requests as if they were each a new request<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=85feac857c659c8ddff3af67099f0b0bd34fb2b2">[85feac857c]</a> Mark Michelson -- Add stateful PJSIP response API call, and use it for out-of-dialog responses.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24840">ASTERISK-24840</a>: res_pjsip: conflicting endpoint identifiers<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94fe4a91781e8f21936cb382c8ecedf1bead44c1">[94fe4a9178]</a> Kevin Harwell -- res_pjsip: Allow configuration of endpoint identifier query order</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f428f25f0ffdfbde3bb4f5bec4e0a219023e6f0">[1f428f25f0]</a> Kevin Harwell -- res_pjsip: Allow configuration of endpoint identifier query order</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0497b7b1550a50cfe6da101f0253fec8866c499c">[0497b7b155]</a> Kevin Harwell -- Revert - res_pjsip: Allow configuration of endpoint identifier query order</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=110b99646cec9128d9dec4bc764dfb585b3e9844">[110b99646c]</a> Kevin Harwell -- res_pjsip: Allow configuration of endpoint identifier query order</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24872">ASTERISK-24872</a>: [patch] AMI PJSIPShowEndpoint closes AMI connection on error<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3292230b87b1e80411dcbb346203b3d8d3ccd5e">[a3292230b8]</a> Richard Mudgett -- chan_pjsip: AMI action PJSIPShowEndpoint closes AMI connection on error.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24755">ASTERISK-24755</a>: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cae712d986211c636f669101bf8071975570fbcf">[cae712d986]</a> Richard Mudgett -- res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24499">ASTERISK-24499</a>: Need more explicit debug when PJSIP dialstring is invalid<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0d018e249177b755cafc9cf8d1fbd6e53839df7">[f0d018e249]</a> Joshua Colp -- res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24685">ASTERISK-24685</a>: "pjsip show version" CLI command<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3044cbf02c2f727fc8cda87ed64acbaafce11ef">[a3044cbf02]</a> Joshua Colp -- res_pjsip: Add "pjsip show version" CLI command.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24727">ASTERISK-24727</a>: PJSIP: Crash experienced during multi-Asterisk transfer scenario.<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d797f17c5a2ad20830073a5fcec28feaecbefd4">[4d797f17c5]</a> Richard Mudgett -- res_pjsip_session: Fix double re-INVITE collision crash.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24741">ASTERISK-24741</a>: dtls_handler causes Asterisk to crash<br/>Reported by: Zane Conkle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8ec15a9ef019d0dcfa875a5f08b9e1277fe701d">[e8ec15a9ef]</a> Kevin Harwell -- res_pjsip: dtls_handler causes Asterisk to crash</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24748">ASTERISK-24748</a>: res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f2eb1931a34f29f5f0430329113fb7f5576ceea">[2f2eb1931a]</a> Joshua Colp -- sorcery: Don't try to load object types which haven't been defined.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24485">ASTERISK-24485</a>: res_pjsip cannot be unloaded or shutdown<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e3d316dd13970370f76e7e19982c8f571f3290b">[9e3d316dd1]</a> Corey Farrell -- res_pjsip: make it unloadable (take 2)</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77a036bf3f23c67bb57bd93a348c1bf468d6e1c5">[77a036bf3f]</a> Corey Farrell -- res_pjsip: make it unloadable</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24615">ASTERISK-24615</a>: When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE<br/>Reported by: David Justl<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=643b81d98e9c8a8f55116c543e8a727ea597b080">[643b81d98e]</a> Joshua Colp -- res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing information on UAS sessions.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24342">ASTERISK-24342</a>: PJSIP: Qualifying endpoints attempts to do them all at the same time.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=64581d894d570a3cfa5ae8b947f326e6fcaa9ef3">[64581d894d]</a> Kinsey Moore -- PJSIP: Stagger outbound qualifies</li>
</ul><br><h4>Category: Resources/res_pjsip_exten_state</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24716">ASTERISK-24716</a>: Improve pjsip log messages for presence subscription failure<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c6e3ad3b4f5fd65578c1381c1f55fdf70d2572b">[8c6e3ad3b4]</a> Joshua Colp -- res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension.</li>
</ul><br><h4>Category: Resources/res_pjsip_messaging</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24937">ASTERISK-24937</a>: [patch]res_pjsip_messaging: Messages may be sent out of order<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ee8424f27780ac3f1e96018e01622d081b640f2">[1ee8424f27]</a> Mark Michelson -- res_pjsip_messaging: Serialize outbound SIP MESSAGEs</li>
</ul><br><h4>Category: Resources/res_pjsip_mwi</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25180">ASTERISK-25180</a>: res_pjsip_mwi: Unsolicited MWI requires reload<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=80e82dc97f85ce55bbdb311ea2dce641df388c70">[80e82dc97f]</a> Joshua Colp -- res_pjsip_mwi: Set up unsolicited MWI upon registration.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24982">ASTERISK-24982</a>: res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b5711683341761b34364ed0c40d521799921921">[7b57116833]</a> Joshua Colp -- res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers.</li>
</ul><br><h4>Category: Resources/res_pjsip_nat</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25830">ASTERISK-25830</a>: Revision 2451d4e breaks NAT<br/>Reported by: Sean Bright<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6f0d7ce9db9a35c8a15a77406f62cf364595e9a9">[6f0d7ce9db]</a> gtjoseph -- config_transport: Fix objects returned by ast_sip_get_transport_states</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25387">ASTERISK-25387</a>: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1dd0e220bf98ca93b825d7b5af4160f7718eab38">[1dd0e220bf]</a> Matt Jordan -- res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25196">ASTERISK-25196</a>: res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=24eec5a10b43c7642ac555b75ed05b054b5e51df">[24eec5a10b]</a> Mark Michelson -- res_pjsip_nat: Adjust when contact should be rewritten.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=028fa546201658ee0c91bc159363e8240ea06067">[028fa54620]</a> Mark Michelson -- res_pjsip_nat: Rewrite route set when required.</li>
</ul><br><h4>Category: Resources/res_pjsip_notify</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25590">ASTERISK-25590</a>: CLI Usage info for 'pjsip send notify' references incorrect config<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b75f587d159cb68ed24b6ee1007ed062f669d79f">[b75f587d15]</a> Corey Farrell -- res_pjsip_notify: Fix CLI usage info</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25737">ASTERISK-25737</a>: res_pjsip_outbound_registration: line option not in Alembic<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=953d1cc11a1b28f8a43f4d68d39f845e47db2d68">[953d1cc11a]</a> gtjoseph -- pjsip/alembic: Add missing columns to system and registration</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25575">ASTERISK-25575</a>: res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f71263e72268bb4966fa7d8f68a0a8b99419ec5">[8f71263e72]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Apply configuration on object type load</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25485">ASTERISK-25485</a>: res_pjsip_outbound_registration: registration stops due to 400 response<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c58091737da86e17cbad4d86ebf4f04055e505fa">[c58091737d]</a> Kevin Harwell -- res_pjsip_outbound_registration: registration stops due to fatal 4xx response</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24907">ASTERISK-24907</a>: res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0ec461a637ecfdd641cd9a9ce62b766472acde46">[0ec461a637]</a> Richard Mudgett -- res_pjsip_outbound_registration.c: Add a serializer shutdown group.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=84c12f9e0c810c4816444dbd2bb8a6f4e5bfc1f9">[84c12f9e0c]</a> Richard Mudgett -- threadpool, res_pjsip: Add serializer group shutdown API calls.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=602c4b74b500fb6fbe3ae3f6e13d2502edbdd56c">[602c4b74b5]</a> Richard Mudgett -- res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c6a95a9ac605c53d1a5863528ff940221684ea3">[8c6a95a9ac]</a> Richard Mudgett -- res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20f3d77ab9cfa7f16c7d34956c660d302a71bc53">[20f3d77ab9]</a> Richard Mudgett -- sorcery: Add ast_sorcery_object_unregister() API call.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4313f32969bc727d667712bba574d7eb875e5b05">[4313f32969]</a> Richard Mudgett -- res_pjsip_outbound_registration.c: Reorder load_module() and unload_module().</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25037">ASTERISK-25037</a>: res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e332c7ed5ebb24e13981d7471f0dd54f3bdd83e6">[e332c7ed5e]</a> Joshua Colp -- res_pjsip_outbound_registration: Fix double unref on error return.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24729">ASTERISK-24729</a>: Outbound registration not occuring on new registrations after reload.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88fbe4e91764320e865a8a39b309821697d84b53">[88fbe4e917]</a> Richard Mudgett -- res_pjsip_outbound_registration: Fix reload race condition.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24514">ASTERISK-24514</a>: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=664067e3189b29bd00276725d2293363c9b9ebe1">[664067e318]</a> Kevin Harwell -- res_pjsip_outbound_publish: stack overflow when using non-default sorcery wizard</li>
</ul><br><h4>Category: Resources/res_pjsip_publish_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24635">ASTERISK-24635</a>: PJSIP outbound PUBLISH crashes when no response is ever received<br/>Reported by: Marco Paland<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6583b4de98dfa8ba1964dc4e685b02d260a749ed">[6583b4de98]</a> Kevin Harwell -- res_pjsip_outbound_publish: eventually crashes when no response is ever received</li>
</ul><br><h4>Category: Resources/res_pjsip_pubsub</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25738">ASTERISK-25738</a>: res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1c4f2a920db173412b38aab785ba22c2cc489f89">[1c4f2a920d]</a> Joshua Colp -- res_pjsip_pubsub: Move where the subscription is stored to after initialized.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25513">ASTERISK-25513</a>: Crash: malloc failed with high load of subscriptions.<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6fbffe42e13d82eebd5545de9a74b6a36bd9a558">[6fbffe42e1]</a> Mark Michelson -- res_pjsip: Set threadpool max size default to 50.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25505">ASTERISK-25505</a>: res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a021a42adaee95d115aa3200467943fecd1f13a">[9a021a42ad]</a> Joshua Colp -- res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25306">ASTERISK-25306</a>: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c126afe18f9073f3ee74e45f574da421131b9fa2">[c126afe18f]</a> Richard Mudgett -- res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e25569ef95c6de6e9267df4673bd1d774b82a000">[e25569ef95]</a> Mark Michelson -- res_pjsip_pubsub: More accurately persist packet.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25057">ASTERISK-25057</a>: res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d649d682c4d28c1f22b8c106d25d0ad982caeebb">[d649d682c4]</a> Joshua Colp -- res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24970">ASTERISK-24970</a>: Crash in res_pjsip_pubsub handling of failed notify<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d4ce7cc2b87317005588e700b278a8cca7005c8">[8d4ce7cc2b]</a> Scott Griepentrog -- res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced</li>
</ul><br><h4>Category: Resources/res_pjsip_refer</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25814">ASTERISK-25814</a>: Segfault at f ip in res_pjsip_refer.so<br/>Reported by: Sergio Medina Toledo<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b1b8e382a46fccd12bd60145cee2ffcf69663a6">[2b1b8e382a]</a> Sergio Medina Toledo -- res_pjsip_refer.c: Fix seg fault in process of Refer-to header.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25204">ASTERISK-25204</a>: res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs.<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05a2cc129396df9d75d1dbafc040eca117f982ba">[05a2cc1293]</a> Mark Michelson -- res_pjsip_refer: Prevent sending duplicate headers.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24700">ASTERISK-24700</a>: CRASH: NULL channel is being passed to ast_bridge_transfer_attended()<br/>Reported by: Zane Conkle<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6d3fcfc3c2f49b3909b7ae0ebb74d99e2fedbb65">[6d3fcfc3c2]</a> Richard Mudgett -- res_pjsip_refer: Fix crash from a REFER and BYE collision.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24376">ASTERISK-24376</a>: res_pjsip_refer: REFER request for remote session attempts to direct channel to external_replaces extension instead of context, without providing for the Referred-To SIP URI<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ea8dd036fa79f7884e71b0d1769d9416a3d958a">[9ea8dd036f]</a> Mark Michelson -- Fix ability to perform a remote attended transfer with PJSIP.</li>
</ul><br><h4>Category: Resources/res_pjsip_registrar</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24785">ASTERISK-24785</a>: 'Expires' header missing from 200 OK on REGISTER<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a507ae31a8e1d890733ddbbf0657de3fd708b86">[7a507ae31a]</a> Joshua Colp -- res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER.</li>
</ul><br><h4>Category: Resources/res_pjsip_sdp_rtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25632">ASTERISK-25632</a>: res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed<br/>Reported by: Olivier Krief<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=219c204a418cbc82ca529837de53cb332ada6b37">[219c204a41]</a> gtjoseph -- pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25356">ASTERISK-25356</a>: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b1561f4c854c37691bd24227b8f722d1dac4291">[1b1561f4c8]</a> Joshua Colp -- res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24769">ASTERISK-24769</a>: res_pjsip_sdp_rtp: Local ICE candidates leaked<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=675b2b8103da5551fcf19398cb0793b8602d4452">[675b2b8103]</a> Matt Jordan -- res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP</li>
</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25297">ASTERISK-25297</a>: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13eb491e35ae6a99164dec6a62d7f05784c75c11">[13eb491e35]</a> Richard Mudgett -- res_pjsip_session.c: Fix crashes seen when call cancelled.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25131">ASTERISK-25131</a>: chan_pjsip: In-dialog authentication not handled.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe21f2e52f1b6629a254cc2f34345c4de6ec4293">[fe21f2e52f]</a> Richard Mudgett -- res_pjsip_session: Fix in-dialog authentication.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25086">ASTERISK-25086</a>: [patch]PJSIP crashes if endpoint missing in Dial()<br/>Reported by: snuffy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f9114179e6bc727cd9fe593d50313c19b6e63492">[f9114179e6]</a> snuffy -- chan_pjsip: Fix crash during off-nominal when no endpoint specified.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24731">ASTERISK-24731</a>: res_pjsip_session cannot be unloaded<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d0df545a443cb00c75fdf1e61db909facd266aa9">[d0df545a44]</a> Corey Farrell -- res_pjsip: Enable unload of all modules at shutdown.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24607">ASTERISK-24607</a>: res_pjsip_session: re-INVITE with declined media streams results in 488<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61fe4f10d230d09b5246c97675637810ffed2fa2">[61fe4f10d2]</a> Joshua Colp -- res_pjsip_session: Fix issue where a declined media stream in a re-INVITE would fail SDP negotiation.</li>
</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25582">ASTERISK-25582</a>: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=18a323e54298fee27d91ef0fe634895960a732ed">[18a323e542]</a> Richard Mudgett -- chan_sip.c: Fix T.38 issues caused by leaving a bridge.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=263a39f2ccf54a313f5ba3f148551b9208caf183">[263a39f2cc]</a> Richard Mudgett -- res_pjsip_t38.c: Back out part of an earlier fix attempt.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=221422be50380259e9b40d53bf029e67b9f19cd7">[221422be50]</a> Richard Mudgett -- bridge core: Add owed T.38 terminate when channel leaves a bridge.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a5bc64491b34bca979c151b4996d7eee8fe1bcf">[0a5bc64491]</a> Richard Mudgett -- channel api: Create is_t38_active accessor functions.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=513638a5f41fe61572477ac59823883a92285272">[513638a5f4]</a> Richard Mudgett -- bridge_channel: Don't settle owed events on an optimization.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c4495cb700019108294877a7ec1935379f77627">[7c4495cb70]</a> Richard Mudgett -- channel.c: Route all control frames to a channel through the same code.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6614babea27fbafbe11820ea03737dd5c4f9ecec">[6614babea2]</a> Matt Jordan -- bridges/bridge_t38: Add a bridging module for managing T.38 state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4875e5ac32f5ccad51add6a4216947bfb385245d">[4875e5ac32]</a> Matt Jordan -- chan_pjsip: Handle T.38 faxes with direct media bridges</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24928">ASTERISK-24928</a>: [patch]t38_udptl_maxdatagram in pjsip.conf not honored<br/>Reported by: Juergen Spies<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cf7d0bf011069fce2e304d76d59771d89e1401e">[4cf7d0bf01]</a> Juergen Spies -- res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24933">ASTERISK-24933</a>: T38 fails negotiation<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f21b45db4920debe736b710951b4860fb420f306">[f21b45db49]</a> Jonathan Rose -- res_pjsip_t38: Fix FAX failures when using PJSIP with authentication</li>
</ul><br><h4>Category: Resources/res_pjsip_transport_websocket</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24106">ASTERISK-24106</a>: WebSockets Automatically decides what driver it will use <br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0393bd6bed3b87b312d7fc252c4fa3782df8260a">[0393bd6bed]</a> Corey Farrell -- chan_sip: Allow websockets to be disabled.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25122">ASTERISK-25122</a>: Large SIP packet received via pjsip over websocket crashes Asterisk <br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=554bd1e39c704a20226c1f8573fe30a327e9ae98">[554bd1e39c]</a> Ivan Poddubny -- res_pjsip_transport_websocket: Fix crash on receiving large SIP packets</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25317">ASTERISK-25317</a>: asterisk sends too many stun requests<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d228b62fd437e02c0638684c1f44c92e5f1e3948">[d228b62fd4]</a> gtjoseph -- stasis_cache_pattern: Backport to 13</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24146">ASTERISK-24146</a>: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec<br/>Reported by: Aleksei Kulakov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=28d924307940700ce2321572b016fdd8263ac7ad">[28d9243079]</a> Eugene Voityuk -- chan_sip.c: Start ICE negotiation when response is sent or received.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25451">ASTERISK-25451</a>: Broken video - erased rtp marker bit<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1435aa3fad5bda73a66dbccf3982787eff55ea2">[a1435aa3fa]</a> Stefan Engström -- res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25438">ASTERISK-25438</a>: res_rtp_asterisk: ICE role message even when ICE is not enabled<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=56ed7b9dd560e468be31684e56a8070b88ae0205">[56ed7b9dd5]</a> Joshua Colp -- res_rtp_asterisk: Move "Set role" warning to be debug.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25265">ASTERISK-25265</a>: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1<br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9a12804e592b97d74ff7b909e0d0022f1ca72386">[9a12804e59]</a> Joshua Colp -- res_rtp_asterisk: Don't leak temporary key when enabling PFS.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aed068844c1c9748da9c67b74ea4d90622be8f46">[aed068844c]</a> Mark Duncan -- res/res_rtp_asterisk: Add ECDH support</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25103">ASTERISK-25103</a>: Roundup - investigate Asterisk DTLS crashes<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ff1ac8797a479ae5416d7c51a761552ecde011e">[7ff1ac8797]</a> Joshua Colp -- res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=55137c3d123626e3a4621bd325b36b62e634abc5">[55137c3d12]</a> Joshua Colp -- res/res_http_websocket: Don't send HTTP response fragmented.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-22805">ASTERISK-22805</a>: res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP <br/>Reported by: Dmitry Burilov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24651">ASTERISK-24651</a>: [patch] Fix race condition in DTLS<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24832">ASTERISK-24832</a>: [patch]DTLS-crashes within openssl <br/>Reported by: Stefan Engström<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25127">ASTERISK-25127</a>: DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending<br/>Reported by: Dade Brandon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25022">ASTERISK-25022</a>: Memory leak setting up DTLS/SRTP calls<br/>Reported by: Steve Davies<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4e207e27ef4c5db85c88969b2aec999e1897ae4">[d4e207e27e]</a> Matt Jordan -- main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b6410c4f880d8f715621de2c95b07ea3489d853">[0b6410c4f8]</a> Steve Davies -- res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24791">ASTERISK-24791</a>: Crash in ast_rtcp_write_report<br/>Reported by: JoshE<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=562b7bf6f09d9ea5ac8e20575d87f4e892609c20">[562b7bf6f0]</a> Matt Jordan -- res/res_rtp_asterisk: Fix crash in debug from RTCP reports without report block</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24337">ASTERISK-24337</a>: Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b22c833c12d69a84fd6e6d33f4529fbc6e37dd47">[b22c833c12]</a> Richard Mudgett -- chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=439e6e1c5d2eaecb4ea132e2c11bb13c42201ab4">[439e6e1c5d]</a> Joshua Colp -- media: Fix crash when determining sample count of a frame during shutdown.</li>
</ul><br><h4>Category: Resources/res_security_log</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-20744">ASTERISK-20744</a>: [patch] Security event logging does not work over syslog<br/>Reported by: Michael Keuter<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e2be8fb8f0df9d622184a49e973ca9173436367">[4e2be8fb8f]</a> Michael L. Young -- main/syslog: Allow dynamic logs, such as security events, to log to the syslog</li>
</ul><br><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24550">ASTERISK-24550</a>: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake<br/>Reported by: Osaulenko Alexander<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05e8e1498207055374312f43fb80fcf0b9d5ab45">[05e8e14982]</a> Joshua Colp -- res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context.</li>
</ul><br><h4>Category: Resources/res_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25882">ASTERISK-25882</a>: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7b6c4decd31377f5d250a41e52d300bd5d1b092d">[7b6c4decd3]</a> Richard Mudgett -- res_stasis: Fix crash on a hanging up channel.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25709">ASTERISK-25709</a>: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel<br/>Reported by: Mark Michelson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eedd77fda03a657375d106427bcd3838fa90ad86">[eedd77fda0]</a> Mark Michelson -- Stasis: Use control queue to prevent crash.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24782">ASTERISK-24782</a>: StasisEnd event not present for channel that was swapped out for another after completing attended transfer<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97ee0ee6c6d5f37591183339999d8cb936bf517a">[97ee0ee6c6]</a> Kevin Harwell -- bridge.c: Fixed race condition during attended transfer</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=35a99b639474f9140fc294c184bb8f0afb1936cf">[35a99b6394]</a> Kevin Harwell -- bridge.c: Hangup attended transfer target if bridged</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d754f70239921561449884e85bd9794b5f515cd9">[d754f70239]</a> Kevin Harwell -- bridge.c: Hangup attended transfer target after it has been swapped out</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24755">ASTERISK-24755</a>: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cae712d986211c636f669101bf8071975570fbcf">[cae712d986]</a> Richard Mudgett -- res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24701">ASTERISK-24701</a>: Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e64d151fae43babc138640adaa59b3b0b2eadb28">[e64d151fae]</a> Kevin Harwell -- ari_websockets: removed extra check on websocket session read</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=72e5ba2ce8ffe6076430add012f56eabaaa2381b">[72e5ba2ce8]</a> Kevin Harwell -- res_http_websocket: websocket write timeout fails to fully disconnect</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24649">ASTERISK-24649</a>: Pushing of channel into bridge fails; Stasis fails to get app name<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f7d23dfcc62c5491956570614365eb7ca39a7655">[f7d23dfcc6]</a> Scott Griepentrog -- stasis transfer: fix stasis bridge push race part two</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=355eb9d22f95c7c435f61922da84feac3b42d5e4">[355eb9d22f]</a> Richard Mudgett -- Bridge core: Pass a ref with the swap channel when joining a bridge.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bdfdb01bcfc830fc8337925a162a5dbb01fddb31">[bdfdb01bcf]</a> Scott Griepentrog -- stasis transfer: fix a race condition on stasis bridge push</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24637">ASTERISK-24637</a>: Channel re-enters Stasis() when it should not<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0d522dbb9a7ef4d364b156f1b74c5d9a38d847">[2b0d522dbb]</a> Scott Griepentrog -- app_bridge: return to the next dialplan priority</li>
</ul><br><h4>Category: Resources/res_stasis_snoop</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24938">ASTERISK-24938</a>: ARI Snoop Channel results in excessive escalating CPU usage<br/>Reported by: George Ladoff<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=68513e00f781e4b9e87a8464b725b96050b3c9c3">[68513e00f7]</a> Kevin Harwell -- res_stasis_snoop: Spying on a single direction continually increases CPU</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25595">ASTERISK-25595</a>: Unescaped : in messge sent to statsd<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9184fbeb347d1168add1f3140af3b6837c8d78db">[9184fbeb34]</a> gtjoseph -- res_pjsip: Use a MD5 hash for static Contact IDs</li>
</ul><br><h4>Category: Resources/res_timing_kqueue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19277">ASTERISK-19277</a>: [patch]endlessly repeating error: "poll failed: Bad file descriptor"<br/>Reported by: Barry Chern<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4dd9560cf6a50621172c34c2d9887041aaf8a3a">[f4dd9560cf]</a> Walter Doekes -- res_timing: Don't close FD 0 when out of open files.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24857">ASTERISK-24857</a>: [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x<br/>Reported by: Justin T. Gibbs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6e6f5b3a1f77f583aa08d387cdcec41a8a7908aa">[6e6f5b3a1f]</a> Justin T. Gibbs -- res/res_timing_kqueue: Update the module to conform to current timer API</li>
</ul><br><h4>Category: Resources/res_timing_pthread</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24768">ASTERISK-24768</a>: res_timing_pthread: file descriptor leak<br/>Reported by: Matthias Urlichs<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ce70587ba61f834af81e81bdee52badcdc493f18">[ce70587ba6]</a> Matthias Urlichs -- res_timing_pthread: Fix leaky pipes.</li>
</ul><br><h4>Category: Resources/res_timing_timerfd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19277">ASTERISK-19277</a>: [patch]endlessly repeating error: "poll failed: Bad file descriptor"<br/>Reported by: Barry Chern<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f4dd9560cf6a50621172c34c2d9887041aaf8a3a">[f4dd9560cf]</a> Walter Doekes -- res_timing: Don't close FD 0 when out of open files.</li>
</ul><br><h4>Category: Resources/res_xmpp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24780">ASTERISK-24780</a>: [patch] - Buddies are always auto-registered when processing the roster<br/>Reported by: Simon Arlott<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=05de9082a599df079ab1f1eb307b60d04a5c3e3b">[05de9082a5]</a> Simon Arlott -- res_xmpp: Buddies are always auto-registered when processing the roster</li>
</ul><br><h4>Category: Tests/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25685">ASTERISK-25685</a>: infrastructure: Run alembic in Jenkins build script<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ff945ab8717a784d4a54d74458733fe2bbb8f2f">[6ff945ab87]</a> Corey Farrell -- Build System: Add support for checking alembic branches.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25611">ASTERISK-25611</a>: core: threadpool thread_timeout_thrash unit test sporadically failing<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b5c13c1545c069d31efd9af709c9f2af85c585a9">[b5c13c1545]</a> Joshua Colp -- test_threadpool: Wait for each task to complete and fix memory leak.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25053">ASTERISK-25053</a>: Unit test category /main/presence missing trailing slash.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=040d2f8558b0775b95fbd60b9a3dc958bb11197b">[040d2f8558]</a> Corey Farrell -- main/test.c: Add test to verify there were no registration errors.</li>
</ul><br><h4>Category: Tests/testsuite</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25582">ASTERISK-25582</a>: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=18a323e54298fee27d91ef0fe634895960a732ed">[18a323e542]</a> Richard Mudgett -- chan_sip.c: Fix T.38 issues caused by leaving a bridge.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=263a39f2ccf54a313f5ba3f148551b9208caf183">[263a39f2cc]</a> Richard Mudgett -- res_pjsip_t38.c: Back out part of an earlier fix attempt.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=221422be50380259e9b40d53bf029e67b9f19cd7">[221422be50]</a> Richard Mudgett -- bridge core: Add owed T.38 terminate when channel leaves a bridge.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a5bc64491b34bca979c151b4996d7eee8fe1bcf">[0a5bc64491]</a> Richard Mudgett -- channel api: Create is_t38_active accessor functions.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=513638a5f41fe61572477ac59823883a92285272">[513638a5f4]</a> Richard Mudgett -- bridge_channel: Don't settle owed events on an optimization.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c4495cb700019108294877a7ec1935379f77627">[7c4495cb70]</a> Richard Mudgett -- channel.c: Route all control frames to a channel through the same code.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6614babea27fbafbe11820ea03737dd5c4f9ecec">[6614babea2]</a> Matt Jordan -- bridges/bridge_t38: Add a bridging module for managing T.38 state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4875e5ac32f5ccad51add6a4216947bfb385245d">[4875e5ac32]</a> Matt Jordan -- chan_pjsip: Handle T.38 faxes with direct media bridges</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25165">ASTERISK-25165</a>: Testsuite - Sorcery memory cache leaks<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc45f4040df77fda402a50822f027f870d114913">[fc45f4040d]</a> Richard Mudgett -- res_sorcery_realtime.c: Fix crash from NULL sorcery object type.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=156395e743189649280066c1497292bb97ed022d">[156395e743]</a> Mark Michelson -- res_sorcery_realtime: Fix leak of sorcery object type.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25318">ASTERISK-25318</a>: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2c73190825bf4c9cedb1031327199767a4a3ca8">[c2c7319082]</a> Joshua Colp -- res_pjsip_session: Don't invoke session supplements twice for BYE requests.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25292">ASTERISK-25292</a>: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=10ba72a9279c591e800ad2656d367b881f73203d">[10ba72a927]</a> Mark Michelson -- Add a test event for inband ringing.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25172">ASTERISK-25172</a>: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e99e654d75a2428ce4b8bc504acf2ec1927779ed">[e99e654d75]</a> Joshua Colp -- app_dial: Hold reference to calling channel formats when dialing outbound.</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25337">ASTERISK-25337</a>: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub<br/>Reported by: Jacques Peacock<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34c64707d1aa346fb0e9c7f97e375d22dedf67d9">[34c64707d1]</a> gtjoseph -- res_pjsip_caller_id: Fix segfault when replacing rpid or pai header</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25615">ASTERISK-25615</a>: res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=36097a185db00230a89f019b9b8ee2d478cc6665">[36097a185d]</a> Richard Mudgett -- Fix sscanf() format string type mismatch.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5b867fa9043dec7aee8fbe21a6537efb103e4d92">[5b867fa904]</a> gtjoseph -- pjsip/config_transport: Check pjproject version at runtime for async ops</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e03582a1c293c0ed7e37896758be613e3e281bfd">[e03582a1c2]</a> gtjoseph -- res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25513">ASTERISK-25513</a>: Crash: malloc failed with high load of subscriptions.<br/>Reported by: John Bigelow<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6fbffe42e13d82eebd5545de9a74b6a36bd9a558">[6fbffe42e1]</a> Mark Michelson -- res_pjsip: Set threadpool max size default to 50.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24963">ASTERISK-24963</a>: ASAN: heap-use-after-free with PJSIP and WSS<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8af6c9cf6bc9f8217fd96f59a6f248330583bdb9">[8af6c9cf6b]</a> Ivan Poddubny -- res_pjsip_transport_websocket: Fix use-after-free bugs.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25018">ASTERISK-25018</a>: pjsip show endpoints crashes asterisk when qualified aors present<br/>Reported by: Ivan Poddubny<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99fb87ae13a1fc28bd5043600def3b373f85a0f6">[99fb87ae13]</a> gtjoseph -- res_pjsip: Fix SEGV on pending-qualify contacts</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24807">ASTERISK-24807</a>: Missing mandatory field Max-Forwards<br/>Reported by: Anatoli<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c24a294f0bd57b297d27e0f4c9e84f14727b5944">[c24a294f0b]</a> Richard Mudgett -- res_pjsip: Fix pjsip.conf type=global object default value handling.</li>
</ul><br><h3>Improvement</h3><h4>Category: Applications/app_stasis</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24802">ASTERISK-24802</a>: stasis: set a channel variable on websocket disconnect error<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7293ecd90b6ac88fda2908ddb48289070e15fd8f">[7293ecd90b]</a> Ashley Sanders -- stasis: set a channel variable on websocket disconnect error</li>
</ul><br><h4>Category: Applications/app_voicemail</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24790">ASTERISK-24790</a>: Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context<br/>Reported by: Graham Barnett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c40d78c31e2a805653e815a7d45da5e5db557ec3">[c40d78c31e]</a> Graham Barnett -- apps/app_voicemail: Demote an ERROR message to a WARNING message</li>
</ul><br><h4>Category: CDR/cdr_manager</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24671">ASTERISK-24671</a>: Missing docs for the CDR AMI Event<br/>Reported by: Dan Jenkins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=452f0eeb57ba8f6f9fc4737fa96a05a2b0d8e91b">[452f0eeb57]</a> Matt Jordan -- AMI: Add documentation for the missing Cdr/CEL events.</li>
</ul><br><h4>Category: CEL/cel_pgsql</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24965">ASTERISK-24965</a>: cel_pgsql - log_error string references CDR instead of CEL<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=abd56db3e07d8177a124958dea412669450b67fa">[abd56db3e0]</a> Rodrigo Ramírez Norambuena -- cel_pgsql: Fix name string for log on unable allocate memory.</li>
</ul><br><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24706">ASTERISK-24706</a>: [patch]add auto-dtmf mode for pjsip<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e76a6a97bf486fd8fbba5ff355e5790d8aa59b85">[e76a6a97bf]</a> Matt Jordan -- contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88b0fa77555b6216d751e156f7f0fdfe33fa9638">[88b0fa7755]</a> yaron nahum -- res_pjsip: Add an 'auto' option for DTMF Mode</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24862">ASTERISK-24862</a>: [patch] Support in-dialog OPTIONS<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2679d0100af28d47bd320b1e7045bde707517789">[2679d0100a]</a> yaron nahum -- res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests</li>
</ul><br><h4>Category: Channels/chan_sip/TCP-TLS</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25043">ASTERISK-25043</a>: [patch] Avoiding ERR_remove_state in OpenSSL<br/>Reported by: Alexander Traud<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2115f11b5402fdea68f42220eb71dca316b19b74">[2115f11b54]</a> Alexander Traud -- tcptls: Avoiding ERR_remove_state in OpenSSL.</li>
</ul><br><h4>Category: Contrib/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25495">ASTERISK-25495</a>: [patch] Prevent old-update packages on repository Debian systems<br/>Reported by: Rodrigo Ramirez Norambuena<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88240f98d999c5fdc68d12a5de1da82684ee98eb">[88240f98d9]</a> Rodrigo Ramírez Norambuena -- install_prereq: Update repositories before install on Debian systems</li>
</ul><br><h4>Category: Core/BuildSystem</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24718">ASTERISK-24718</a>: [patch]Add inital support of "sanitize" to configure<br/>Reported by: Badalian Vyacheslav<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=486b172b50ae5b525d03ea7467bdb4ffa7ad90fd">[486b172b50]</a> Ivan Poddubny -- Build: Add menuselect options for using compiler sanitizers</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24133">ASTERISK-24133</a>: [patch]Please support Clang; Allow no-exec stacks<br/>Reported by: Jeffrey Walton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f5bc032567ac1af93e8d5320a7a6c126a8639611">[f5bc032567]</a> Diederik de Groot -- Add support for the clang compiler; update RAII_VAR to use BlocksRuntime</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24813">ASTERISK-24813</a>: asterisk.c: #if statement in listener() confuses code folding editors<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ac00c6bc2dfe89d61b52c988e38c8765c698ef59">[ac00c6bc2d]</a> Corey Farrell -- main/asterisk.c: Reverse #if statement in listener() to fix code folding.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25518">ASTERISK-25518</a>: taskprocessor: Add high water mark<br/>Reported by: Jonathan Rose<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ff48319d9a3d0e4dd301f90d4b9b214f9f87e3a">[6ff48319d9]</a> Jonathan Rose -- taskprocessor: Add high water mark warnings</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25310">ASTERISK-25310</a>: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED<br/>Reported by: Guido Falsi<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ed9c9a280c08a17fce602c15d2b01de199ca736">[4ed9c9a280]</a> Guido Falsi -- Core/General: Add #ifdef needed on FreeBSD.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25256">ASTERISK-25256</a>: [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.<br/>Reported by: Richard Mudgett<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=875aee4c09a1780ac57b38fbb74a7bec2503fba0">[875aee4c09]</a> Richard Mudgett -- pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25045">ASTERISK-25045</a>: vector: Add new capabilities and unit tests<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f9aea8e3ceeabcde699001ab5e0547405a5ba78">[5f9aea8e3c]</a> gtjoseph -- vector: Additional enhancements and fixes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7a7e9733c2288e255f6c3bcc2a56f7088e08b834">[7a7e9733c2]</a> gtjoseph -- vector: Traversal, retrieval, insert and locking enhancements</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25051">ASTERISK-25051</a>: Remove unneeded uses of optional_api providers.<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad6ea2969706c1d19290d2db32957830e73a75ae">[ad6ea29697]</a> Corey Farrell -- Remove unneeded uses of optional_api providers.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24917">ASTERISK-24917</a>: [patch] clang compilation warnings<br/>Reported by: Diederik de Groot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9c3ed428759a83a2b80106e605fe43dda0569425">[9c3ed42875]</a> Diederik de Groot -- Update configure.ac/Makefile for clang</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e9788056e9f60a8ef6f810f0c5d31e0f3700e3bb">[e9788056e9]</a> Matt Jordan -- channels/chan_skinny: Fix compilation error introduced in f8e21a1adf</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cb318f39605f06e814af00965fe33f7d8b70ccff">[cb318f3960]</a> Diederik de Groot -- Example script for scan-build (the llvm static analyzer)</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bb16bedc72d5cf6fc95cda3bd60c440285575d7">[1bb16bedc7]</a> Diederik de Groot -- Clang: change previous tautological-compare fixes.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d6dfc85666370e1bc71f050bd5dfaf12cb629734">[d6dfc85666]</a> Diederik de Groot -- Clang: Fix some more tautological-compare warnings.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2be9cc26434fa91dc6f5bf9a39097d0ef4582a13">[2be9cc2643]</a> Diederik de Groot -- Fix/Update clang-RAII macro implementation</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d0d78d573256ae0b89545c26b5966c9662395395">[d0d78d5732]</a> Diederik de Groot -- clang compiler warnings: Fix various warnings for tests</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ba6e3dffd57f1c62c0c7f9a9030c1f41b5a6350">[6ba6e3dffd]</a> Diederik de Groot -- clang compiler warnings: Fix autological comparisons</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f324870dab2f13a0314bd7203dad52392b6089bf">[f324870dab]</a> Diederik de Groot -- clang compiler warnings: Fix pointer-bool-converesion warnings</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=488f093e97b52a1337c7f480aac16f27bf3908b1">[488f093e97]</a> Diederik de Groot -- clang compiler warnings: Fix sometimes-initialized warning in func_math</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c027133f6d564027ccbbfbbd21eddb9d17368439">[c027133f6d]</a> Diederik de Groot -- clang compiler warnings: Fix non-literal-null-conversion warnings</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d54ccda3b18aaee4857bccec82ed462c8854009f">[d54ccda3b1]</a> Diederik de Groot -- clang compiler warnings: Remove large chunks of unused code from extconf</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0ecd472e4f890cb932265028bca9b4c2fef562ea">[0ecd472e4f]</a> Diederik de Groot -- clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e7be5b2dc03ff5685b92408e303caeeb68c7598">[4e7be5b2dc]</a> Diederik de Groot -- clang compiler warnings: Fix format specified in framehook</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f8faf16af107855e6afa2574b3e9436ed20250c">[5f8faf16af]</a> Diederik de Groot -- clang compiler warnings: Fix -Wabsolute-value warnings</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09b681e344b5f40544ae0de8685a4010cba1d21e">[09b681e344]</a> Diederik de Groot -- clang compiler warnings: Fix invalid enum conversion</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f33abb827135082ccc1528bc4e01206f847f468">[7f33abb827]</a> Matt Jordan -- main/stdtime/localtime: Fix warning introduced in r433720</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=47eeb67e147d26298d1350ea539bb9666259d6aa">[47eeb67e14]</a> Diederik de Groot -- clang compiler warnings: Ignore -Wunused-command-line-argument</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dbb4d6f9e7a820d3ab95550fc4d392c97c098f2f">[dbb4d6f9e7]</a> Diederik de Groot -- clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e126ab9eebc6db005df31280ff8e596ba6562426">[e126ab9eeb]</a> Diederik de Groot -- clang compiler warnings: Fix a variety of "unused" warnings</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f6534527deef4dcc40001b93601761cf643a7c4">[2f6534527d]</a> Diederik de Groot -- clang compiler warnings: Fix -Wself-assign</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb70993a50c5c85b61d7b1cf9e4a7d6e5eb9a901">[eb70993a50]</a> Diederik de Groot -- clang compiler warnings: Fix -Wparantheses-equality warnings</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0ff16036a6619e819db95199e619505696f5556">[c0ff16036a]</a> Diederik de Groot -- clang compiler warnings: Fix -Wbitfield-constant-conversion warning</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=844bc76bef50be6bdeaf10010d9024ffd6aca948">[844bc76bef]</a> Diederik de Groot -- clang compiler warnings: Fix -Winitializer-overrides</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e204042d9f24083de56e267ad752abf16820f97">[5e204042d9]</a> Diederik de Groot -- clang compiler warnings: Fix -Wunused-function; make inline function static</li>
</ul><br><h4>Category: Core/HTTP</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24316">ASTERISK-24316</a>: For httpd server, need option to define server name for security purposes<br/>Reported by: Andrew Nagy<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=112d23c73e0a81f5870ba9f58aa588b89ae31c44">[112d23c73e]</a> Ashley Sanders -- HTTP: For httpd server, need option to define server name for security purposes</li>
</ul><br><h4>Category: Core/ManagerInterface</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8237bd357d8fc633e16b09bdeb34f0bc20801ea7">[8237bd357d]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4a05879d652ffeb7f7e0be6d9122221bd68da04">[d4a05879d6]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f21f85c3733ff633937549c3ba8983df0636dce">[2f21f85c37]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
</ul><br><h4>Category: Core/PBX</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25040">ASTERISK-25040</a>: pbx: Improve performance of reloads by making hint destruction more performant<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=399cd8bcd9e53f30d4a36b67200281407f27798e">[399cd8bcd9]</a> Matt Jordan -- main/pbx: Resolve case sensitivity regression in PBX hints</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1b19c15f17d7275b7f807e0532b9a5d2ea829aa2">[1b19c15f17]</a> Matt Jordan -- main/pbx: Improve performance of dialplan reloads with a large number of hints</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25067">ASTERISK-25067</a>: Sorcery Caching: Implement a new caching module<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b74b07136973b2c753acff02f6e88326de7a5ef0">[b74b071369]</a> Joshua Colp -- res_sorcery_memory_cache: Backport to 13</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25044">ASTERISK-25044</a>: sorcery: Add ability to insert a new wizard into an object type's list<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=637c8f065efead83f56455a4ab45785e34ed56fb">[637c8f065e]</a> gtjoseph -- sorcery: Add API to insert/remove a wizard to/from an object type's list</li>
</ul><br><h4>Category: Documentation</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24892">ASTERISK-24892</a>: Super Awesome Company sound prompts<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4dbd4021c91a6a612746ed5e116021e270bc6ff5">[4dbd4021c9]</a> Rusty Newton -- configs/basic-pbx: Modified main IVR to play new Allison prompt.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24671">ASTERISK-24671</a>: Missing docs for the CDR AMI Event<br/>Reported by: Dan Jenkins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=452f0eeb57ba8f6f9fc4737fa96a05a2b0d8e91b">[452f0eeb57]</a> Matt Jordan -- AMI: Add documentation for the missing Cdr/CEL events.</li>
</ul><br><h4>Category: Features</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24678">ASTERISK-24678</a>: [PATCH] Added atxfer* settings to features.conf.sample<br/>Reported by: Niklas Larsson<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d606d87bfb7eeacda9fcc3575f9383d29731d2e">[7d606d87bf]</a> Niklas Larsson -- configs/samples/features.conf.sample: Document attended transfer DTMF options</li>
</ul><br><h4>Category: Resources/res_ari</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24802">ASTERISK-24802</a>: stasis: set a channel variable on websocket disconnect error<br/>Reported by: Kevin Harwell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7293ecd90b6ac88fda2908ddb48289070e15fd8f">[7293ecd90b]</a> Ashley Sanders -- stasis: set a channel variable on websocket disconnect error</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8237bd357d8fc633e16b09bdeb34f0bc20801ea7">[8237bd357d]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4a05879d652ffeb7f7e0be6d9122221bd68da04">[d4a05879d6]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f21f85c3733ff633937549c3ba8983df0636dce">[2f21f85c37]</a> Kevin Harwell -- ARI/AMI: Include language in standard channel snapshot output</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74b032bb0341a260c8147b86b00901284e4f5e56">[74b032bb03]</a> Joshua Colp -- ari: Add support for specifying an originator channel when originating.</li>
</ul><br><h4>Category: Resources/res_ari_applications</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</li>
</ul><br><h4>Category: Resources/res_ari_bridges</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24870">ASTERISK-24870</a>: ARI: Subscriptions to bridges generally not super useful<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90165e306d958293bae47dd901e2c672dca95006">[90165e306d]</a> Matt Jordan -- res/res_stasis: Fix accidental subscription to 'all' bridge topic</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b50e372394bf0950ebbc96793d9594de97282749">[b50e372394]</a> Matt Jordan -- ARI: Add events for Contact and Peer Status changes</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3502c0431db52d00eb16dc1cc2462be7a509ba5e">[3502c0431d]</a> Matt Jordan -- res/res_stasis_device_state: Allow for subscribing to 'all' device state</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c9f613309d66ae6a8e5454cd53276459bcd2674">[4c9f613309]</a> Matt Jordan -- ARI: Add the ability to subscribe to all events</li>
</ul><br><h4>Category: Resources/res_ari_channels</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24412">ASTERISK-24412</a>: [patch]Incomplete channel originate/continue handling with ARI<br/>Reported by: Nir Simionovich (GreenfieldTech - Israel)<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42b342c6e265680209d74b634bf6e71062ccb682">[42b342c6e2]</a> Mark Michelson -- Add the ability to continue and originate using priority labels.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74b032bb0341a260c8147b86b00901284e4f5e56">[74b032bb03]</a> Joshua Colp -- ari: Add support for specifying an originator channel when originating.</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25618">ASTERISK-25618</a>: res_pjsip: Check for readability of TLS files at startup<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=21962dad93fdb887899676597779a6ae47ff1edb">[21962dad93]</a> gtjoseph -- res_pjsip: Add existence and readablity checks for tls related files</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25477">ASTERISK-25477</a>: pjsip show "command" like [criteria]<br/>Reported by: Bryant Zimmerman<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=162acd45f744395c19ec5686af30d0abd61ef897">[162acd45f7]</a> gtjoseph -- res_pjsip: Add "like" processing to pjsip list and show commands</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25114">ASTERISK-25114</a>: res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes<br/>Reported by: George Joseph<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=262d590819b123b1f57196beef8aca45c4aa0d09">[262d590819]</a> gtjoseph -- res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24706">ASTERISK-24706</a>: [patch]add auto-dtmf mode for pjsip<br/>Reported by: yaron nahum<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e76a6a97bf486fd8fbba5ff355e5790d8aa59b85">[e76a6a97bf]</a> Matt Jordan -- contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=88b0fa77555b6216d751e156f7f0fdfe33fa9638">[88b0fa7755]</a> yaron nahum -- res_pjsip: Add an 'auto' option for DTMF Mode</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24918">ASTERISK-24918</a>: pjsip: add CLI options to display global and system configuration<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5737650a672ff2b1e2fdd8b76de345d12ede81c8">[5737650a67]</a> Kevin Harwell -- res_pjsip: add CLI command to show global and system configuration</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24575">ASTERISK-24575</a>: [patch]Make capath work for res_pjsip<br/>Reported by: cloos<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bc4a89e1f9cb71295e074158b938e5d800342b0">[8bc4a89e1f]</a> cloos -- Add support for the ca_list_path option for PJSIP transports.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24643">ASTERISK-24643</a>: res_pjsip: Add user=phone option<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b521c612fc2116cc381412c8e72bf8a719922481">[b521c612fc]</a> Matt Jordan -- res_pjsip: Backport missing commits for user_eq_phone</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=006ffdcfb252c358f2831ef239cebe71a084467f">[006ffdcfb2]</a> Matt Jordan -- res_pjsip: Add 'user_eq_phone' option to add a 'user=phone' parameter when applicable.</li>
</ul><br><h4>Category: Resources/res_pjsip_caller_id</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25791">ASTERISK-25791</a>: res_pjsip_caller_id: Lack of support for Anonymous <anonymous@anonymous.invalid><br/>Reported by: Anthony Messina<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27f32cd0a69141daf44c2cc6c9c0e52b37af4a16">[27f32cd0a6]</a> gtjoseph -- res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited</li>
</ul><br><h4>Category: Resources/res_pjsip_keepalive</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24644">ASTERISK-24644</a>: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=915bb88d3e973f647eb9d9e560688d6a02af2c2a">[915bb88d3e]</a> Matt Jordan -- res_pjsip_keepalive: Add runtime configurable keepalive module for connection-oriented transports.</li>
</ul><br><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25072">ASTERISK-25072</a>: res_pjsip_outbound_registration: line functionality. Additional check for using the request URI<br/>Reported by: Dmitriy Serov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42476e66333a9b9841b56b2207760a70b1b835d1">[42476e6633]</a> demon-ru -- res_pjsip_outbound_registration: Check request URI for line.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24949">ASTERISK-24949</a>: res_pjsip_outbound_registration: Backport line functionality<br/>Reported by: Joshua Colp<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89f6719f7abb775f4db4a772b358bb0c7075a52d">[89f6719f7a]</a> Joshua Colp -- res_pjsip_outbound_registration: Add virtual line support.</li>
</ul><br><h4>Category: Resources/res_pjsip_publish_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24811">ASTERISK-24811</a>: asterisk-publication sorcery object does not use realtime<br/>Reported by: Matt Hoskins<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e806f9e1293eb56cdf0163f2585a1aa36da7376">[8e806f9e12]</a> Matt Hoskins -- ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk.</li>
</ul><br><h4>Category: Resources/res_statsd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25572">ASTERISK-25572</a>: Endpoints: Add StatsD stats for Asterisk endpoints<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d27aac0a9d4b7b72ddb73ae45f6f7327110a07dc">[d27aac0a9d]</a> Matt Jordan -- res/res_endpoint_stats: Add module to emit endpoint StatsD statistics</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25571">ASTERISK-25571</a>: PJSIP: Add StatsD stats for some common PJSIP objects<br/>Reported by: Matt Jordan<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=90d9a70789a0874cff3a29caca5046995a54dbd4">[90d9a70789]</a> Matt Jordan -- res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75097a0955ca707ac8f6dc0d4def9b9d3b9c2b8a">[75097a0955]</a> Matt Jordan -- res/res_pjsip_outbound_registration: Add registration statistics for StatsD</li>
</ul><br><h4>Category: Sounds</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25068">ASTERISK-25068</a>: Move commonly used FreePBX extra sounds to the core set<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b2c8a99f9ed6b18f1aca489a4eb6c66f2dda83ea">[b2c8a99f9e]</a> Rusty Newton -- sounds/Makefile: Incremented core and extra sounds versions to 1.5</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24892">ASTERISK-24892</a>: Super Awesome Company sound prompts<br/>Reported by: Rusty Newton<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4dbd4021c91a6a612746ed5e116021e270bc6ff5">[4dbd4021c9]</a> Rusty Newton -- configs/basic-pbx: Modified main IVR to play new Allison prompt.</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24744">ASTERISK-24744</a>: Swedish Core Voice prompts<br/>Reported by: Tove Hjelm<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=394fcb5eab5b1be1a346746c69a29d700ad91494">[394fcb5eab]</a> Rusty Newton -- sounds: Add Swedish sounds to Makefile and XML</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24575">ASTERISK-24575</a>: [patch]Make capath work for res_pjsip<br/>Reported by: cloos<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8bc4a89e1f9cb71295e074158b938e5d800342b0">[8bc4a89e1f]</a> cloos -- Add support for the ca_list_path option for PJSIP transports.</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>New Feature</h3><h4>Category: Channels/chan_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24341">ASTERISK-24341</a>: PJSIP Ability to get info per contact<br/>Reported by: xrobau<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a7c38428af5da34b99334ad384976c1330c9b569">[a7c38428af]</a> Joshua Colp -- pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.</li>
</ul><br><h3>Bug</h3><h4>Category: Addons/chan_ooh323</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25299">ASTERISK-25299</a>: RTP port leaks with incoming OOH323 calls<br/>Reported by: Alexandr Dranchuk<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=480c443e2691272a7227e0949244e80e53bc31b2">[480c443e26]</a> Alexander Anikin -- chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy</li>
</ul><br><h4>Category: Applications/app_amd</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25639">ASTERISK-25639</a>: app_amd: system maxwords discrepency<br/>Reported by: Dade Brandon<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1d3d20dd685e2b5896adabbaa833f56dc49abd69">[1d3d20dd68]</a> Dade Brandon -- app_amd: Correct documentation to reflect functionality</li>
</ul><br><h4>Category: Applications/app_queue</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-19820">ASTERISK-19820</a>: wrapuptime is intermittently disregarded for queue calls<br/>Reported by: WRP<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=338a8ffed673e4c3a828c7c216575f8e3e712350">[338a8ffed6]</a> Martin Tomec -- app_queue: Add member flag "in_call" to prevent reading wrong lastcall time</li>
</ul><br><h4>Category: CDR/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25458">ASTERISK-25458</a>: Unable to set CDR variable in h extension or hangup_handler<br/>Reported by: Ross Beer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1087b0c6eda6a2372f923a6cac6dccc865fd438a">[1087b0c6ed]</a> Matt Jordan -- main/cdr: Allow setting properties on a finalized CDR if it is the last one</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1f23e65b89c939779c109689d6e99d79b01cd2bf">[1f23e65b89]</a> Matt Jordan -- main/cdr: Set the end time on a CDR if endbeforehexten is Yes</li>
</ul><br><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25307">ASTERISK-25307</a>: Hangup on channel using FastAGI does not hang up child channels<br/>Reported by: David Cunningham<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=80a8b2a4cd34f8bd02f3edd07af9316f562463c9">[80a8b2a4cd]</a> Richard Mudgett -- app_dial: Immediately exit dial if the caller is already hung up.</li>
</ul><br><h4>Category: Core/Sorcery</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25141">ASTERISK-25141</a>: pjsip_options: Contact reference leak<br/>Reported by: Corey Farrell<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5dc9fb4198f2081f3996c89fb42aaffc0f326df8">[5dc9fb4198]</a> gtjoseph -- res_pjsip/location: Fix ref leak in contact_apply_handler</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9e7827e3ac057f22bc17823a44778b76270c5901">[9e7827e3ac]</a> Corey Farrell -- pjsip_configuration: Fix leak in persistent_endpoint_update_state.</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=857166b5e5dbdb81b4c25a42f36842a394989768">[857166b5e5]</a> gtjoseph -- res_pjsip/location: Fix memory leak in permanent_uri_handler</li>
</ul><br><h4>Category: General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25323">ASTERISK-25323</a>: Asterisk: ongoing segfaults uncovered by CHAOS_DEBUG<br/>Reported by: Scott Griepentrog<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1ea7a5a774ee7c70325e1be37afb403dac1abbc3">[1ea7a5a774]</a> Scott Griepentrog -- CHAOS: cleanup possible null vars on msg alloc failure</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c37c7071f82eca8cc068fee6c16a842eddbddb6">[3c37c7071f]</a> Scott Griepentrog -- CHAOS: prevent crash on failed strdup</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c94f46080f60435fffd197d14441ccf9d963521b">[c94f46080f]</a> Scott Griepentrog -- CHAOS: avoid crash if string create fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cc59533b903b3d55e8b388f28385287e712ae62">[4cc59533b9]</a> Richard Mudgett -- CHAOS: res_pjsip_diversion avoid crash if allocation fails</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fb6b5c684b8772ba008339a417725a208f72409e">[fb6b5c684b]</a> Scott Griepentrog -- PJSIP: avoid crash when getting rtp peer</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f72f9ceefca52002c45f5910219dbcb0f9437a79">[f72f9ceefc]</a> Scott Griepentrog -- pjsip: avoid possible crash req_caps allocation failure</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6862c2a167f4ed2cb8511bb1ae94a13582afa25b">[6862c2a167]</a> Scott Griepentrog -- Chaos: handle failed allocation in get_media_encryption_type</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1cd6366588c66dce5be66541ceb7f828fde3773">[f1cd636658]</a> Scott Griepentrog -- Chaos: make hangup NULL tolerant</li>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab373f2ceffcad3a497663027199f4f4a81f644b">[ab373f2cef]</a> Scott Griepentrog -- CHAOS: prevent sorcery object with null id</li>
</ul><br><h4>Category: PBX/pbx_spool</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-17069">ASTERISK-17069</a>: Callfile retries behave erratically as file size grows<br/>Reported by: Jeremy Kister<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d5864a358c0adf813ed0122eb32006b43fe688de">[d5864a358c]</a> Ivan Poddubny -- pbx/pbx_spool: Fix issue when call files were executed too early</li>
</ul><br><h4>Category: Resources/res_hep_rtcp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25352">ASTERISK-25352</a>: res_hep_rtcp correlation_id is different then res_hep<br/>Reported by: Kevin Scott Adams<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78d0b9d97ecf034468e252440217dd4bc371ef71">[78d0b9d97e]</a> Matt Jordan -- channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id</li>
</ul><br><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25229">ASTERISK-25229</a>: Exchanging Device and Mailbox State Using PJSIP fails after restart of peer<br/>Reported by: Vadim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f295088764ba25f1234b874337bd9363cc7d1ed1">[f295088764]</a> Alexei Gradinari -- res_pjsip_outbound_publish: Fix processing 412 response</li>
</ul><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25689">ASTERISK-25689</a>: pjsip show contacts not working in Asterisk 13.7rc2<br/>Reported by: Marcelo Terres<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cd58c3b20e04e1acd3d77c964dc1dd9a7b85d74">[4cd58c3b20]</a> Mark Michelson -- res_sorcery_realtime: Remove leading ^ requirement.</li>
</ul><br><h4>Category: Resources/res_pjsip_publish_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25229">ASTERISK-25229</a>: Exchanging Device and Mailbox State Using PJSIP fails after restart of peer<br/>Reported by: Vadim<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f295088764ba25f1234b874337bd9363cc7d1ed1">[f295088764]</a> Alexei Gradinari -- res_pjsip_outbound_publish: Fix processing 412 response</li>
</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25645">ASTERISK-25645</a>: res_rtp_asterisk: Lock inversion<br/>Reported by: Steve Davies<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a160cdbf6382c2ff7de0cc0396214e14e49f955">[3a160cdbf6]</a> Joshua Colp -- res_rtp_asterisk: Revert DTLS negotiation changes.</li>
</ul><br><h4>Category: Resources/res_xmpp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25735">ASTERISK-25735</a>: [patch] res_xmpp: Does not connect in component mode<br/>Reported by: Karsten Wemheuer<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0405c31756e45047cec20caccf228e5862550f9c">[0405c31756]</a> Karsten Wemheuer -- res_xmpp: Does not connect in component mode</li>
</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-24602">ASTERISK-24602</a>: Unable to call WebRTC client via wss on chan_pjsip<br/>Reported by: Oleg Kozlov<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d32e516c7cd48979db092a82b97a7ac4a743f526">[d32e516c7c]</a> Martin Tomec -- res/pjsip: Mark WSS transport as secure</li>
</ul><br><h3>Improvement</h3><h4>Category: Core/General</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-25627">ASTERISK-25627</a>: Easily Preventable Compile Warning<br/>Reported by: Diederik de Groot<ul>
<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4285dee7783c0bbfbda79a59f07734b9d16d1aa1">[4285dee778]</a> Diederik de Groot -- include/asterisk/time.h: Renamed global declaration:tv</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=454daec0e1916ab4d51514a6b22d9aff6a6c504c">454daec0e1</a></td><td>Joshua Colp</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ba2b5e92ccae5bac400255cc25eb8a11c78f05b">4ba2b5e92c</a></td><td>Joshua Colp</td><td>.version: Update for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e6f27ca09c0308ad802237aecb3598adb2f78df4">e6f27ca09c</a></td><td>Joshua Colp</td><td>.lastclean: Update for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=08dbdd59960d29aaef8c45bd700c67658121cd00">08dbdd5996</a></td><td>Joshua Colp</td><td>realtime: Add database scripts for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec7a89771d021a1a4d697cad46206d39d21edb32">ec7a89771d</a></td><td>Joshua Colp</td><td>ChangeLog: Updated for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ffcb6512054fd14bb75217c74077b4434f6594cc">ffcb651205</a></td><td>Joshua Colp</td><td>Release summaries: Add summaries for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=97499f717adfdf40d777ec968069d95d0d23745d">97499f717a</a></td><td>Joshua Colp</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99d52771b55b1cfffe5ecc1b602fe0b9c25087b8">99d52771b5</a></td><td>Joshua Colp</td><td>.version: Update for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb9e193c6542a7efd7d89cd9846e09126617c220">eb9e193c65</a></td><td>Joshua Colp</td><td>.lastclean: Update for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8ec588b8b171540c183d31541965f54070cf550d">8ec588b8b1</a></td><td>Joshua Colp</td><td>realtime: Add database scripts for certified/13.8-cert1-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c29e2e3fb7b3b63bb76942ac75844070b7bff037">c29e2e3fb7</a></td><td>Joshua Colp</td><td>.version: Update for certified/13.8</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c796e694e2d413810fa21ee8ff06e61a76c3220">3c796e694e</a></td><td>Matt Jordan</td><td>Disable extended support modules</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fad0410486b3a47743331f14fcb565b79357887c">fad0410486</a></td><td>Mark Michelson</td><td>ChangeLog: Updated for 13.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f885f00766bf73576760ea0dde6f8b635246baf">0f885f0076</a></td><td>Mark Michelson</td><td>Release summaries: Add summaries for 13.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1fa37aebd5881d1baa3f66f53cd3c8ad56081a4">a1fa37aebd</a></td><td>Mark Michelson</td><td>Release summaries: Remove previous versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7de5fd4392ac18e6897735f72b354d8f5c020f7">e7de5fd439</a></td><td>Mark Michelson</td><td>.version: Update for 13.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8baf8138486fb64a9771243dd561099e1616dae8">8baf813848</a></td><td>Mark Michelson</td><td>.lastclean: Update for 13.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42469df2056cb9d8f68ffd0068d9b1ba1c1b662e">42469df205</a></td><td>Mark Michelson</td><td>realtime: Add database scripts for 13.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06f5ace1fa80ce0799c6d25954de2236a1f842c8">06f5ace1fa</a></td><td>Mark Michelson</td><td>ChangeLog: Updated for 13.8.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6984246785c5960989e7a47d6d28df9365e0b09">a698424678</a></td><td>Mark Michelson</td><td>Release summaries: Add summaries for 13.8.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e395a0b973fbcb6acf29eb61e41de5d8e6af0a83">e395a0b973</a></td><td>Mark Michelson</td><td>.version: Update for 13.8.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=38a86b2dbf84a91b1c9fdad9455deeb2629f7fdc">38a86b2dbf</a></td><td>Mark Michelson</td><td>.lastclean: Update for 13.8.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0c8c8bf4a2bb3a9f7ec1d40cdc0d5592f4b24e8">e0c8c8bf4a</a></td><td>Mark Michelson</td><td>realtime: Add database scripts for 13.8.0-rc1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9ae21b510f77899190661e36fe25b07f3fa0f092">9ae21b510f</a></td><td>Richard Mudgett</td><td>chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=56bcb97a3cae238a2ff06db2ff2a46e3dd04599f">56bcb97a3c</a></td><td>Richard Mudgett</td><td>chan_sip.c: Simplify sip_pvt destructor call levels.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=677a65fcbb093864d199feef60b41f56fe532ba9">677a65fcbb</a></td><td>Joshua Colp</td><td>build: Add configure check for proto field of PJSIP TLS transport setting.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32f0a3d52a45e1c59274e271a83722cafbf5f231">32f0a3d52a</a></td><td>gtjoseph</td><td>build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=38499e71256d7ca862a8a1493e427572ca2d8529">38499e7125</a></td><td>gtjoseph</td><td>pjproject: Pass (dont_)optimize flags to pjproject and fix pjsua</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=875d5e9872a04d1d9668eeb2f2534cc1ffdf4204">875d5e9872</a></td><td>gtjoseph</td><td>pjproject_bundled: Remove --with-external-pa from configure options.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3c8076a83b345a4a970909de928d11b0c5b32a0f">3c8076a83b</a></td><td>gtjoseph</td><td>install_prereq: Add packages for bundled pjproject</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7cf7b0a4f97d32de8620391d2ab7da5ab959d243">7cf7b0a4f9</a></td><td>gtjoseph</td><td>third_party/Makefile.rules: Replace unsupported != operator with $(shell ...)</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=53f57001f25aa8bda13d25e7152e35d977aaa86d">53f57001f2</a></td><td>gtjoseph</td><td>loader: Retry dlopen when loading fails</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26b8f2692e43ab382153fbb29c138427994ded92">26b8f2692e</a></td><td>Joshua Colp</td><td>res_pjsip_dtmf_info: NULL terminate the message body.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=86d6e44cc1ebc1c0e4ed5cf872ca9416d276c038">86d6e44cc1</a></td><td>gtjoseph</td><td>alembic: Fix downgrade and tweak for sqlite</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9633be9d258cc6ec1f7e0071d82a6d2cc8605e16">9633be9d25</a></td><td>Richard Mudgett</td><td>func_callerid.c: Update REDIRECTING reason documentation.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4165ea7778be093f614a8c476e04648b846b862e">4165ea7778</a></td><td>Richard Mudgett</td><td>SIP diversion: Fix REDIRECTING(reason) value inconsistencies.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=41f4af4ce535eee2a4a3d0186b9f16f5de825a72">41f4af4ce5</a></td><td>Richard Mudgett</td><td>res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4c5998ff556eed8e9cb266de86c83cde95b103aa">4c5998ff55</a></td><td>Richard Mudgett</td><td>res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b59956a875817367834431e7f1fa02486b5aed7f">b59956a875</a></td><td>gtjoseph</td><td>build-system: Allow building with static pjproject</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ee947d4a7a9679a03e48fde7c1a55adb4a6b0960">ee947d4a7a</a></td><td>gtjoseph</td><td>res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6e70e8ccdb77e772ef877793e88b82daf626db2c">6e70e8ccdb</a></td><td>gtjoseph</td><td>res_sorcery_memory_cache: Fix SEGV in some CLI commands</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4417f64d833e37b975c729f0e57c40e52e21d013">4417f64d83</a></td><td>Leif Madsen</td><td>Add initial support to build Docker images</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e7a6abbbd3fbf7a1ec239dec3b1bdcd7ce4cb0ff">e7a6abbbd3</a></td><td>Richard Mudgett</td><td>rtp_engine.h: Remove extraneous semicolons.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6656afffa0bf19eeaeecf6d775fa4b68f6c92248">6656afffa0</a></td><td>Richard Mudgett</td><td>chan_sip.c: Suppress T.38 SDP c= line if addr is the same.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea9deff996744b41ae383876b6c4e64936d85a9c">ea9deff996</a></td><td>Christof Lauber</td><td>res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d2a1457e0b4ecdd512fe58fdb55ecc07fd141bea">d2a1457e0b</a></td><td>gtjoseph</td><td>res_pjsip/config_transport: Allow reloading transports.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6b921f706d493f926c3bceb44775aa71bdc5e225">6b921f706d</a></td><td>gtjoseph</td><td>res_pjproject: Add ability to map pjproject log levels to Asterisk log levels</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f1f79812c10bd6f1659c4f4f0e3788a80834e435">f1f79812c1</a></td><td>Mark Michelson</td><td>Fix failing threadpool_auto_increment test.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a3a857dd62638609ce0d7dbc8d97287f1aa6580">5a3a857dd6</a></td><td>Richard Mudgett</td><td>cel.c: Fix mismatch in ast_cel_track_event() return type.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=87ab65c557eccfe91c8aea6eba0568aa96de6412">87ab65c557</a></td><td>gtjoseph</td><td>res_odbc: Fix exports.in for missing symbols</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ebe167f79267e5e071911d583df58a42bf5d1153">ebe167f792</a></td><td>Mark Michelson</td><td>Fix creation race of contact_status structures.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b1b797e0e75231e1669f9f7e17cc3032657c24cb">b1b797e0e7</a></td><td>gtjoseph</td><td>res_pjsip: Refactor load_module/unload_module</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e9e896abd1f1abae526319fdfd474ae1c28fed52">e9e896abd1</a></td><td>Badalyan Vyacheslav</td><td>Build: Fix menuselect USAN conflicts</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=93e8ed01544bb1b9cba1dc24b06e157e3add9e23">93e8ed0154</a></td><td>Corey Farrell</td><td>Simplify and fix conditional in FD_SET.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a7c8d4cd6b1db086c9efaa48ace361f8894fe767">a7c8d4cd6b</a></td><td>Joshua Colp</td><td>tests/test_sorcery_memory_cache_thrash: Improve termination process.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6f978fbfe5f2a90244ea6a84bd7063adbe97ba85">6f978fbfe5</a></td><td>Richard Mudgett</td><td>app_confbridge: Only use b_profile options from the conference.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec8fd6714d3f82f3ef3f3f53c0c93285dd7470fe">ec8fd6714d</a></td><td>gtjoseph</td><td>chan_misdn: Fix a few issues causing compile errors</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6a799cd78f0f26f26afab8b123e04f16c0308c36">6a799cd78f</a></td><td>Mark Michelson</td><td>Check for OpenSSL defines before trying to use them.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=23829b325377ae32f78cef81a48cc4318a7206b9">23829b3253</a></td><td>Mark Michelson</td><td>res_stasis_device_state: Fix refcounting error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4e8e6d3922303711335bee677676c4cddc18dd5d">4e8e6d3922</a></td><td>Sean Bright</td><td>res_rtp_asterisk: Allow ICE host candidates to be overriden</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a6ee8caebf023fe8433b9d46fde457c17de5c81">2a6ee8caeb</a></td><td>gtjoseph</td><td>logging: Remove/fix some message annoyances</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8804d0973c9144da4b3af863a24a8dfaac22c04e">8804d0973c</a></td><td>gtjoseph</td><td>build_system: Fix some warnings highlighted by clang</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=109b0aff6bcd73a634f440b6c8ec84a155e4ba66">109b0aff6b</a></td><td>gtjoseph</td><td>res/Makefile: Fix bug in "clean" target for ari</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a85fab7c44b36f03d19d212b49c01b958060432d">a85fab7c44</a></td><td>gtjoseph</td><td>pjsip/alembic: Fix definition of qualify_timeout</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=aa9348ab9aa0069e5992be38321f7849697206f1">aa9348ab9a</a></td><td>Stefan Engström</td><td>chan_sip.c: AMI &amp; CLI notify methods get different values of asterisk's own ip.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=65bd4fcc3f535ac21ec68ec497f3d47a28b5c02a">65bd4fcc3f</a></td><td>Mark Michelson</td><td>res_odbc: Remove connection management</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2a9e623ff94689b149156871e9f91c9db0fb87c0">2a9e623ff9</a></td><td>Richard Mudgett</td><td>config_options.c: Fix warning message wording.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ed3c9c15126ddd2437bed6d022194fe7d1c90c29">ed3c9c1512</a></td><td>Richard Mudgett</td><td>app_confbridge.c: Replace inlined code with existing function.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0d40afa69e78bfd28a8246b995160f960c57eba">f0d40afa69</a></td><td>Richard Mudgett</td><td>app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9da18af992a18c2ea11fef22b300ca203923245b">9da18af992</a></td><td>gtjoseph</td><td>res_pjsip: Add res_pjproject dependency to UPGRADE.txt and samples</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4a3275abb9e6528f5bacbb454446a3e2e7115d88">4a3275abb9</a></td><td>Mark Michelson</td><td>Stasis: Use custom structure when setting variables.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8261bda1bf63d76f0913a5273c9b7ae1729addec">8261bda1bf</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Prevent crash from AMI command on freed subscription.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1003c2eb0505c34c6fe3ebd587fe3adf413b979a">1003c2eb05</a></td><td>Mark Michelson</td><td>Stasis: Fix potential memory leak of control data.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f87c3275cc4d60871167283e621565d51b35ff8d">f87c3275cc</a></td><td>Richard Mudgett</td><td>res_pjsip: Add CLI "pjsip dump endpt [details]"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=137fe5ae010df4ff808fa4aecd3f166c6698ba45">137fe5ae01</a></td><td>gtjoseph</td><td>res_pjproject: Add module providing pjproject logging and utils</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0ab89182d9175cbae6944269d7e8f3e59bce6e1b">0ab89182d9</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Increase CLI "core ping taskprocessor" timeout.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a2a8ea33309e77aaec16fb2ed00f7bdf4e83c4a7">a2a8ea3330</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Fix some taskprocessor unrefs.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d604a9afc875f00288438447a2c0a7d70258f14f">d604a9afc8</a></td><td>Richard Mudgett</td><td>Fix alembic branches on v13.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0c79f3a4fdd7f5db09c257742156cdb4f86ac8e">a0c79f3a4f</a></td><td>gtjoseph</td><td>pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5644bca9f91711db2585e0e96c450e60ade39f90">5644bca9f9</a></td><td>Daniel Journo</td><td>Update version number in features.conf.sample</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9cda1de34dac5fe9afbb1cf2b46edfb1b3083b69">9cda1de34d</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Simplify ast_taskprocessor_get() return code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a79af2b312e4283f1eb44df1ae67c4dd5b58d1c0">a79af2b312</a></td><td>Richard Mudgett</td><td>astmm.c: Add more stats to CLI "memory show" commands.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5586abc95728996ce9e36c54a1ee8f6d8de9c525">5586abc957</a></td><td>Richard Mudgett</td><td>res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts".</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cf8e7a580bd865ebd563b3eba261111a908bd41a">cf8e7a580b</a></td><td>Richard Mudgett</td><td>res_pjsip: Create human friendly serializer names.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4276f185f0718b2d0de2d021c3629cc719c78745">4276f185f0</a></td><td>Richard Mudgett</td><td>Sorcery: Create human friendly serializer names.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f02ac1b7f954a3445da9007b3d9bf4f0706e750a">f02ac1b7f9</a></td><td>Richard Mudgett</td><td>Stasis: Create human friendly taskprocessor/serializer names.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ec1f1c6742d55dac75f715422ce4c0191e0f216e">ec1f1c6742</a></td><td>Richard Mudgett</td><td>taskprocessor.c: New API for human friendly taskprocessor names.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d8bc3e0c8b4daaa806232cb1c7a7e654b88a5c29">d8bc3e0c8b</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Fix CLI "core show taskprocessors" output format.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c4b7502dedcc4358dd4f41cb23001f0fa4075d6">2c4b7502de</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Fix CLI "core show taskprocessors" unref.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b33ac7a468cc351e3453e87d1ebda44a912c6c2">3b33ac7a46</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Sort CLI "core show taskprocessors" output.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fc32c4dd37d4ee3e0807047acad5218ecdc8f67">0fc32c4dd3</a></td><td>Richard Mudgett</td><td>ccss.c: Replace space in taskprocessor name.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0e0c24ad78020b3cf47f69632e2e88c20c0a7387">0e0c24ad78</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0f79c8839b08f4c8d0d7c0aca2e1a3573b148bca">0f79c8839b</a></td><td>Diederik de Groot</td><td>main: Use ast_strdup instead of strdup</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=881dc862e0648f2a63b39028f7007aa37a0926f3">881dc862e0</a></td><td>gtjoseph</td><td>asterisk.h: Add ASTERISK_REGISTER_FILE macro</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e462f0063f3e7fd2a2d8f0603d9c7cf8de46a88f">e462f0063f</a></td><td>Corey Farrell</td><td>main/pbx: Move hangup handler routines to pbx_hangup_handler.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab191d124c5f1ac085c6c1e53a9983f879f194e5">ab191d124c</a></td><td>Corey Farrell</td><td>main/pbx: Move dialplan application management routines to pbx_app.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=09a9b93896a4a027528d98d0447c8e9175fb80ff">09a9b93896</a></td><td>Corey Farrell</td><td>main/pbx: Move switch routines to pbx_switch.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c608274a39f9916ed72d7b3f8f971ac7e8218f32">c608274a39</a></td><td>Corey Farrell</td><td>main/pbx: Move timing routines to pbx_timing.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4ec85a9f076d64d01110da32817e40fb96cd5321">4ec85a9f07</a></td><td>gtjoseph</td><td>voicemail: Move app_voicemail / res_mwi_external conflict to runtime</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fdcfd77247b461bfe7dd38c3a56a12f648534de">7fdcfd7724</a></td><td>Corey Farrell</td><td>main/pbx: Move variable routines to pbx_variables.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2ffade45742a5c4361998625bef58eca0fbdb95e">2ffade4574</a></td><td>Corey Farrell</td><td>main/pbx: Move custom function routines to pbx_functions.c.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=20b8474f20fc387d5b355ee81b751e4988fc28d7">20b8474f20</a></td><td>gtjoseph</td><td>main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a2804007585aca2aa0100d12dfa3413efeb084a5">a280400758</a></td><td>Joshua Colp</td><td>test_time: Provide a timeout when waiting.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=136c537695224b6edbb9ab8550dc33cf6b29eb83">136c537695</a></td><td>Dade Brandon</td><td>res_http_websocket.c: prevent avoidable disconnections caused by write errors</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f2efbb5d7521d28c12f781616813df7af1389555">f2efbb5d75</a></td><td>Corey Farrell</td><td>Remove res_jabber file that was left behind.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dde7f3c1c4c013dc7975c818ee36978e6944faa3">dde7f3c1c4</a></td><td>Matt Jordan</td><td>res_pjsip_history: Add a module that provides PJSIP history for debugging</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=be050f26382b215b9fb3e4b86c13e41929327c71">be050f2638</a></td><td>Dade Brandon</td><td>chan_sip.c: fix websocket_write_timeout default value</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a9941de9d24093b5ff44096d1d7406f29d11e45">0a9941de9d</a></td><td>Matt Jordan</td><td>res/res_pjsip_location: Delete contact_status object when contact is deleted</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1e24a0ca8ae075af2814668fc99ecfabb47423b3">1e24a0ca8a</a></td><td>Kevin Harwell</td><td>res_rtp_asterisk: rtp-&gt;ice check not wrapped in HAVE_PJPROJECT ifdef</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0cefcabd589bf8ecf33988d7c710932ab01037e5">0cefcabd58</a></td><td>Joshua Colp</td><td>rtp_engine: Ignore empty filenames in DTLS configuration.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=158a0a54223c0314c8a4b07ab6982fe5db0942b7">158a0a5422</a></td><td>Joshua Colp</td><td>chan_sip: Enable WebSocket support by default.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=53bd5a539a52c93ebf23a7d0ce892c7a83b91159">53bd5a539a</a></td><td>Mark Michelson</td><td>Alembic: Increase column size of PJSIP AOR "contact".</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=da17dc4d751184b1b358757db63cd9a34222c219">da17dc4d75</a></td><td>Mark Michelson</td><td>Alembic: Add PJSIP global keep_alive_interval.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94f9927784e63d0e30aa7919b83b0e0fcc35c57e">94f9927784</a></td><td>Matt Jordan</td><td>main/utils: Don't emit an ERROR message if the read end of a pipe closes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=529535f0c2dd2fca84e31287dd7a00622cacd3c8">529535f0c2</a></td><td>Matt Jordan</td><td>Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb0b60619daddb1448e980f9067780bcd6ca5e35">bb0b60619d</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Fix off nominal ref leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3fcf160faeb036529c575b66d73e7978f475fb28">3fcf160fae</a></td><td>Niklas Larsson</td><td>CHANGES: Fix a typo</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=59881fbb9988be2f4e07ca750f45a404e79cb115">59881fbb99</a></td><td>David M. Lee</td><td>Fixed some typos</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b94d9a10d5001ddb2c6a9aee4b66ee92ec3a3c8">2b94d9a10d</a></td><td>Matt Jordan</td><td>res/res_pjsip_t38: Add debug statements</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=af288b2d9650bf7fdc30591e82a06b6c7610b80f">af288b2d96</a></td><td>Matt Jordan</td><td>main/cli: Use proper string methods to check existence of context/exten/app</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3354b325c67824b4aa052fb81693d28e792886a6">3354b325c6</a></td><td>Matt Jordan</td><td>res_statsd: Add functions that support variable arguments</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d4a522d587bb1986cc66ed59a087be3784eaaceb">d4a522d587</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e44ab3816cae3e85d27e969366a12881af42fa46">e44ab3816c</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Fix 423 response handling.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f62b642fe32d06ca24b4a0de94543279fe918d0a">f62b642fe3</a></td><td>Matt Jordan</td><td>res/res_pjsip: Fix off nominal crash with requests that fail and have a timer</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c0f2f8de458e68e412be91ccc363a9f7aae77c78">c0f2f8de45</a></td><td>Richard Mudgett</td><td>res_pjsip_rfc3326.c: Fix crash when channel goes away.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f43b85c92050c8deba7041e687404228294d920">4f43b85c92</a></td><td>Mark Michelson</td><td>Taskprocessors: Increase high-water mark</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=367972e42d1a5f73d8bb4abacd4c681fc77dcd24">367972e42d</a></td><td>Mark Michelson</td><td>res_pjsip distributor: Don't send 503 response to responses.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f9cb7d62bf1ee2d3f7d878607d2d1eb9995dd03">2f9cb7d62b</a></td><td>Mark Michelson</td><td>res_pjsip: Deny requests when threadpool queue is backed up.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8410336681b54766f148b7708f08d6d5e7ff6f2e">8410336681</a></td><td>Walter Doekes</td><td>docs: Fix a few typo's in app docs (more then, resourse).</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=afec1b1b6497c2b81c0ef468861933b6ba277562">afec1b1b64</a></td><td>Matt Jordan</td><td>res_pjsip/location: Destroy contact_status objects on contact deletion</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=715f770c9ff011284c1e87f9b5bcde1fc02ab4df">715f770c9f</a></td><td>Matt Jordan</td><td>pjsip_configuration: On delete, remove the persistent version of an endpoint</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f0f190af08c5f7f4af3316209b174bd92be145c3">f0f190af08</a></td><td>Matt Jordan</td><td>main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=11e54b1932b62a084722cb547a51a5fc2ca4d423">11e54b1932</a></td><td>Matt Jordan</td><td>pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=118d628e084311af0096178b096e3959ac603edd">118d628e08</a></td><td>Matt Jordan</td><td>Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ebe69dee0d0132bcda93ce909d0a82316e86e8f7">ebe69dee0d</a></td><td>Mark Michelson</td><td>format_cap: Detect vector allocation failures.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b19efefef0cdc7203611bf9d161766ef6922558">3b19efefef</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a346f095fb5f02391d870d2527a665ea926e65b">0a346f095f</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Ensure dialog lock balance.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad395080957b33a17f6cfe2c83697bebef286c25">ad39508095</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Prevent crashes on final NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=067f40876001255aed9bf8b65567d1c25961aebd">067f408760</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Remove serializer when sending final NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1bcc5927655c71f2ea0db31c0cd0a3bf0095714d">1bcc592765</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Fix crash on destruction of empty subscription tree.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b3cc2bd7dfad379cec77e7333cc93c23fda6aa92">b3cc2bd7df</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Solidify lifetime and ownership of objects.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8c65dfa413cff6ad5af12350564f37d4786fe01">c8c65dfa41</a></td><td>Richard Mudgett</td><td>strings.c: Fix __ast_str_helper() to always return a terminated string.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b271d4a28a607341f2374b6f8200b7f4f775e5e6">b271d4a28a</a></td><td>Richard Mudgett</td><td>Add missing failure checks to ast_str_set_va() callers.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fd2adc20430221dadd58c60d57655b95da168c6">9fd2adc204</a></td><td>Matt Jordan</td><td>rest-api-templates: Wikify error code response reasons</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9fc9777fa34753fb38991d42d8dbed516e907ca2">9fc9777fa3</a></td><td>Matt Jordan</td><td>contrib/scripts/autosupport: Update for Asterisk 13</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e14023ca3543d91bc108f8f21af0509c2a428e47">e14023ca35</a></td><td>Richard Mudgett</td><td>config.c: Fix off-nominal memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a99e821520405be48241f2a51c659cefd6939da2">a99e821520</a></td><td>Richard Mudgett</td><td>config.c: Fix potential memory corruption after [section](+).</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8f777ab58499f6b3b9264c9bd6750e0ad5eb033c">8f777ab584</a></td><td>Debian Amtelco</td><td>chan_pjsip: Add Referred-By header to the PJSIP REFER packet.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ced0a2d71b690f24026392bcfbbe4c36eb8d4dff">ced0a2d71b</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Shutdown in a less crash potential order.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cc279eea11853ad90605775a63d58a1cab88f96c">cc279eea11</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Misc tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9af3b613f6423e73a28546df5808155a9fc3cfa3">9af3b613f6</a></td><td>Richard Mudgett</td><td>res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ddebb217f00c00e95259d3977d6d8b43adc69c65">ddebb217f0</a></td><td>Richard Mudgett</td><td>sched.c: Add warning about negative time interval request.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d30939b6e84f18eb1fa3eb9819951fe8a1c764f4">d30939b6e8</a></td><td>Kevin Harwell</td><td>ARI: Changed version from 1.8.0 to 1.9.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5f19c9baded56a5adb419a7bcb1ac00fbe09f404">5f19c9bade</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Fix user sort compare function.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3a8576403968bd1f49c8bef67735c04d05fb6983">3a85764039</a></td><td>Richard Mudgett</td><td>res/ari/config.c: Optimize conf_alloc() object init.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bbeda190c3f05e95a82c7d9609c66dcf3ce35bd3">bbeda190c3</a></td><td>Richard Mudgett</td><td>app_dial.c: Remove some no-op code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fe5077b1f8caed2df419e1bd7b872657b7def726">fe5077b1f8</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Eliminate race during initial NOTIFY.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5c713fdf18ffa934e0cac8ddb29e4ad95a68200b">5c713fdf18</a></td><td>Mark Michelson</td><td>scheduler: Use queue for allocating sched IDs.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e75aff53e6ee68833595db101e43329adf9a4459">e75aff53e6</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d91d01df180a712485aa60d14fda2aa9e5063d2">4d91d01df1</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add some notification comments.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f36a9d122171781dd8a99d32792a0c19103b1f15">f36a9d1221</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94582f8fabb926d197ce0f3a01208b385975ec09">94582f8fab</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix off-nominal memory leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b3ed52239b24546b1ee12156dadccb70db7403e">8b3ed52239</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Fix one byte buffer overrun error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4329bd1e4c059e714122465901ea2c46dd924b71">4329bd1e4c</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Use ast_alloca() instead of alloca().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a456a20ecf835bcf70ebc2a279e230df402bec08">a456a20ecf</a></td><td>Richard Mudgett</td><td>res_pjsip_pubsub.c: Add missing error return in load_module().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f58f4c6e2762e5ad7eccf7065e63b345f4cda7f6">f58f4c6e27</a></td><td>Richard Mudgett</td><td>res_pjsip/location.c: Use the builtin ao2_callback() match function instead.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4eedd9ef9d7c000cd8d67cbeb1789ac6d71860aa">4eedd9ef9d</a></td><td>Matt Jordan</td><td>main/config_options: Check for existance of internal object before derefing</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=695f26cbb759f8ecc18c6b1d6cf84b3105b2f007">695f26cbb7</a></td><td>David M. Lee</td><td>res_rtp_asterisk: Add more ICE debugging</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=61c6c6aa6c60636207567a49c6320946c1840e99">61c6c6aa6c</a></td><td>David M. Lee</td><td>Fix when remote candidates exceed PJ_ICE_MAX_CAND</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad9cb6c2ce6dbe9c985c6891daf53cc4160e3a13">ad9cb6c2ce</a></td><td>Mark Michelson</td><td>res_pjsip: Fix contact refleak on stateful responses.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7c4d0c3506374b89502bd6c1bda89c3f241b6708">7c4d0c3506</a></td><td>Joshua Colp</td><td>res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0582776f7fe55e72acda586fb32185ad7879aeab">0582776f7f</a></td><td>Richard Mudgett</td><td>ari/ari_websockets.c: Fix ast_debug parameter type mismatch.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=77518d54344945d7d3bfc1ebfe61d97704fa5dfa">77518d5434</a></td><td>Richard Mudgett</td><td>res_http_websocket.c: Fix some off nominal path cleanup.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c61547fee6fdc1e132d359311da48e87d98d25b1">c61547fee6</a></td><td>Richard Mudgett</td><td>res_ari.c: Add missing off nominal unlock and remove a RAII_VAR().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd867cd0787b124984caf2604478212651ea4c03">bd867cd078</a></td><td>Richard Mudgett</td><td>app_queue.c: Extract some functions for simpler code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ded51e3d77a13eed059a85a083d0ab0324a77db7">ded51e3d77</a></td><td>Richard Mudgett</td><td>app_queue.c: Fix error checking in QUEUE_MEMBER() read.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b719f56c72c9cc66879eeef11de2ef4498cba648">b719f56c72</a></td><td>Mark Michelson</td><td>res_pjsip_sdp_rtp: Restore removed NULL check.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cea5dc7b8afd0e8cbde4c5d253bac3219125b168">cea5dc7b8a</a></td><td>Richard Mudgett</td><td>audiohook.c: Simplify variable usage in audiohook_read_frame_both().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e18c300550453df06507e32c4bc78ef91d369f27">e18c300550</a></td><td>Joshua Colp</td><td>res_http_websocket: When shutting down a session don't close closed socket</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e194047acb3ba846356041c1a6222caefc65a2e">8e194047ac</a></td><td>Matt Jordan</td><td>res/res_format_attr_silk: Expose format attributes to other modules</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a0f451c35ed56b08353c4c3150bf847867f74fe7">a0f451c35e</a></td><td>Matt Jordan</td><td>main/format: Add an API call for retrieving format attributes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=26f0559a94519b735396ed4ce90e23c1d9d5b332">26f0559a94</a></td><td>David M. Lee</td><td>Replace htobe64 with htonll</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=27dc2094e98b59b8a50b059ddd6048285a42e6b9">27dc2094e9</a></td><td>Mark Michelson</td><td>res_http_websocket: Debug write lengths.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=39cc28f6ea2140ad6d561fd4c9e9a66f065cecee">39cc28f6ea</a></td><td>Mark Michelson</td><td>res_http_websocket: Avoid passing strlen() to ast_websocket_write().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1519eb44a796911c7c438bbe4e31bb89be244387">1519eb44a7</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Must protect mime_types_len with mime_types_lock.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a93b7a927c5975c0a8889dc66868f81e4eef8aa3">a93b7a927c</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=741fa0d26d38f7a9ed98c595ee1bb6b6ce8a9923">741fa0d26d</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp.c: Fixup some whitespace.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b21fd9a38bcd89402249440c1670ce48781f30">89b21fd9a3</a></td><td>Richard Mudgett</td><td>rtp_engine.h: No sense allowing payload types larger than RFC allows.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7427c7f13b1d0c9095e83a3ea38394f521d3a75e">7427c7f13b</a></td><td>Richard Mudgett</td><td>rtp_engine.c: Minor tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e20f435b6093c2c72e678b9fad1ed037c3191b88">e20f435b60</a></td><td>Richard Mudgett</td><td>rtp_engine.h: Misc comment fixes.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc5d7f9c37fe8a4ff5744ab8620898ccae6a7d2a">bc5d7f9c37</a></td><td>Richard Mudgett</td><td>chan_sip.c: Tweak glue-&gt;update_peer() parameter nil value.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=48698a5e21d7307f61b5fb2bd39fd593bc1423ca">48698a5e21</a></td><td>Mark Michelson</td><td>res_http_websocket: Properly encode 64 bit payload</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f78a4b52b8ed7b5b367c3465652a7ce98fe9175d">f78a4b52b8</a></td><td>Matt Jordan</td><td>Bump the ARI version to 1.8.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4e19e414adfdd1d5eea792ce1a3dc61d5a7874d">b4e19e414a</a></td><td>Mark Michelson</td><td>res_pjsip: Add rtp_keepalive to sample config file.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a23adcca3d747f14590ddf7860f72418a4475866">a23adcca3d</a></td><td>Michael Cargile</td><td>res/res_musiconhold: Add a warning when MOH does not exist</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=03064daeb2bc78339400b0c05f657032a3e27011">03064daeb2</a></td><td>Matt Jordan</td><td>res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=243c0d1609d193130f24d4586070036fdb900f0f">243c0d1609</a></td><td>Richard Mudgett</td><td>parking_applications.c: Fix ast_verb() line terminator.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2735dd5b2dd7cbc9ef2a8cf2573f53338f36fb62">2735dd5b2d</a></td><td>Richard Mudgett</td><td>res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d0ca343ca4c3f81e8739bc5a7aaed12b89a207e">3d0ca343ca</a></td><td>Richard Mudgett</td><td>res_pjsip_session.c: Add some helpful comments and minor tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8d08bb179ce16e52f2340664c3c6aa4fb6432733">8d08bb179c</a></td><td>Richard Mudgett</td><td>res_pjsip_session.c: Fix off nominal crash potential in debug message.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a1a550593b6aa00df9f9e3c3643ba402c4c0193">0a1a550593</a></td><td>Matt Jordan</td><td>apps/app_dictate: Fix typo in attribution</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0b6ff77afbad10b96a31932dd697e06748281d80">0b6ff77afb</a></td><td>Matt Jordan</td><td>res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f0d6d346c83db96b13b0f37d90a8ac48453bbe3">2f0d6d346c</a></td><td>Matt Jordan</td><td>res/res_pjsip_outbound_registration: Fix WARNING message</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cd2213f1ae7ef14b9a7d50984a5dd4b431a8e195">cd2213f1ae</a></td><td>Matt Jordan</td><td>res_pjsip/configuration: Fix a variety of default value problems</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2e4bdbd78adb4df16af1132df8e58f464e039cd4">2e4bdbd78a</a></td><td>Matt Jordan</td><td>main/sorcery: Provide log messages when a wizard does not support an operation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2325b106fd61817d6be65195b5568945cd5e3084">2325b106fd</a></td><td>Matt Jordan</td><td>tests/test_devicestate: Add additional tests for the device state API</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=328f0be806cc8236288c8760007ab149a865a196">328f0be806</a></td><td>Matt Jordan</td><td>main/devicestate: Prevent duplicate registration of device state providers</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bee41eec6291d11cb47e7cda1b5abbedb44dccfc">bee41eec62</a></td><td>Matt Jordan</td><td>res/res_sorcery_memory_cache: Fix test registration issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d738e9026055665bbf7ce99dd03ace1782a674e">4d738e9026</a></td><td>Matt Jordan</td><td>tests/test_sorcery_memory_cache_thrash: Fix test loading problems</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=47ea312b2402bf31a0f1ccb9d73a6b8b7d814b09">47ea312b24</a></td><td>Benjamin Ford</td><td>ARI: Added new functionality to get all module information.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=38bace4fbb8d2c47cc38230645de2aa0769726fb">38bace4fbb</a></td><td>Richard Mudgett</td><td>res_pjsip_t38.c: Fix always false if test.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2f7688c7881d79f8ac1deb311766208f28468cf2">2f7688c788</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=74be3a50d79b7b76f6c567e63c3adc7d3f1f135d">74be3a50d7</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Eliminate a simple RAII_VAR.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=589e93617a77e2523bda05f5b275d4501a072cd7">589e93617a</a></td><td>Richard Mudgett</td><td>res_pjsip_mwi.c: Fix mid-line log message line breaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49f81ddb85a4460ae69569a87a0ea1ae264e3019">49f81ddb85</a></td><td>Matt Jordan</td><td>Makefile: Remove coverage files on 'make clean'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=78a1f4aa4683702f5c5747b677a7944cb94866c3">78a1f4aa46</a></td><td>Richard Mudgett</td><td>chan_vpb.cc: Fix compiler warning Jenkins found.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8e07ab145d7e4085968696a17fc7167d8ee2ad06">8e07ab145d</a></td><td>Matt Jordan</td><td>sorcery/realtime: Add a bit of debug and warning messages for bad configs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5e9c4e9b2faf10aef7635879e2a5cce1be9119b">a5e9c4e9b2</a></td><td>Matt Jordan</td><td>res/res_corosync: Always decline module load, instead of failing</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2602a7484be67353ac70691e7e75a54983d4d773">2602a7484b</a></td><td>Richard Mudgett</td><td>test.c: Add unit test registration checks for summary and description.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b0482d699e44d1f4159bd68d30b2a6a80d21ca8">2b0482d699</a></td><td>Richard Mudgett</td><td>Unit tests: Fix unit test description strings.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=036bc0012f1af3b41268a9f8d4cbbdb11f8995bc">036bc0012f</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Add missing line endings to CLI commands</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bec7435945282ef8d646f2a63494995286bebb9f">bec7435945</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c2519fdf1c8ff99e966c3e868f75305cd3826475">c2519fdf1c</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Misc code cleanups.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a2b718f4f625c02da3d36546d680836dfb8ef9bd">a2b718f4f6</a></td><td>Richard Mudgett</td><td>res_pjsip.h: Fix some doxygen comments.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=32ddf6d86b9f47a044f293eea2401b5a34c68595">32ddf6d86b</a></td><td>Richard Mudgett</td><td>taskprocessor.c: Remove extra unref from off-nominal path.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0090216db3e6d4b7796bbb0b230225bfdbe068e">e0090216db</a></td><td>ibercom</td><td>CLI: Cosmetic issue - core show uptime</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d908272b7e03d8b5002e4d3d83bcf8725c57afdd">d908272b7e</a></td><td>David M. Lee</td><td>Fixes for OS X</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1558a891293dd96ba71a3cedf0caa1968067ba5e">1558a89129</a></td><td>gtjoseph</td><td>Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change"</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5a42397018f8b850b3901de0a09b8befad602b37">5a42397018</a></td><td>Joshua Colp</td><td>sorcery: Fix cache creation callback.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=51ffed5e6123c0f2b816067ebbd27b9dbd85d2d5">51ffed5e61</a></td><td>Matt Jordan</td><td>res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7950b65e4f707d36cb8d17e82288e913c4d86f22">7950b65e4f</a></td><td>Matt Jordan</td><td>res/res_pjsip_exten_state: Fix confusing NOTICE message</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d8a462356a938eea82e8424242d89a682495b57">9d8a462356</a></td><td>Matt Jordan</td><td>ARI: Update version to 1.7.0</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7fcf0a97b897c958de83ee49fe403c52c8a5d3ed">7fcf0a97b8</a></td><td>gtjoseph</td><td>app_playback: Suppress warnings on playback if channel hung up</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9b13536feda1e9a723106a0092662f468de12672">9b13536fed</a></td><td>Rodrigo Ramírez Norambuena</td><td>main/manager.c: Bugfix sort action_manager by alphabetically</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16f602f5c2297f2d75360efe27893df00b34285a">16f602f5c2</a></td><td>Yousf Ateya</td><td>res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6553a00770dad889928324a99702ab5dee0fd38b">6553a00770</a></td><td>Rodrigo Ramírez Norambuena</td><td>cdr_pgsql: Use PQescapeStringConn for escaping names.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ea917fefafd878f4021e5e7a929b848d8032f28e">ea917fefaf</a></td><td>gtjoseph</td><td>vector: Add REMOVE, ADD_SORTED and RESET macros</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=613a461c3dd7a51ec0673c3c925d3fa0129e4765">613a461c3d</a></td><td>Sean Bright</td><td>res_rtp_asterisk: Issue ERROR if res_srtp is not found.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5392e970d0653d686e02332cc04901564a1ea181">5392e970d0</a></td><td>gtjoseph</td><td>doc: Make progdocs play nice with git</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=608f0a94eeb7bcee3947742c58d2d6218d3e10bb">608f0a94ee</a></td><td>Ivan Poddubny</td><td>contrib/editors: Fix vim syntax highlighting of comments in config files</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8b0f85ac06d4ebbb8684e5c857724a3d1accde64">8b0f85ac06</a></td><td>gtjoseph</td><td>test_vector: Fix build breakage caused by ASTERISK_REGISTER_FILE</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=525c8c8689225008f0916266cb06eae98dde45bf">525c8c8689</a></td><td>Rodrigo Ramírez Norambuena</td><td>include/asterisk/channel.h: Fix typo</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3efe0df044c9e186205b16aa5b1ea8f9cc0ed92b">3efe0df044</a></td><td>Corey Farrell</td><td>Sample Configs: Fix syntax error in pjsip.conf</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b8cddfb36288723fd6b65b7e65d15da2bbd2d41">4b8cddfb36</a></td><td>Mark Michelson</td><td>res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=415a0d0745f8ecb0c05e993e0b28b9e79f982cdf">415a0d0745</a></td><td>Joshua Colp</td><td>res_ari_device_states: Fix dependency on res_stasis_device_state.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49ef81c15c114001f727129e9cfc2a4acfe2c5fa">49ef81c15c</a></td><td>Joshua Colp</td><td>res_sorcery_config: Fix build issue due to syntax error.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=df23c8a86bf8b24d4a9d9c5d02665a1fd391c90f">df23c8a86b</a></td><td>Joshua Colp</td><td>res_pjsip_outbound_registration: Fix build due to removal of transaction.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b0e929219b40f134a925d68dbd764910e3389f3e">b0e929219b</a></td><td>gtjoseph</td><td>.gitignore: Add .gcno and .gcda</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3327560cb2e0ce43f55c0f1ca6a712c5bfe625dc">3327560cb2</a></td><td>Mark Michelson</td><td>res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b74b2cdcda5b5c2228e55142ec33d3c16790e233">b74b2cdcda</a></td><td>gtjoseph</td><td>pjsip_options: Fix format specifier for int64_t rtt.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5d218cde872ed3fa2dc5ad92ed3a17b413ce8203">5d218cde87</a></td><td>gtjoseph</td><td>More .gitignore updates</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7d43d85bea6cba63b4473d7655d268e67c82bde8">7d43d85bea</a></td><td>gtjoseph</td><td>.gitignore updates for master/13</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3d27c223a5bb2b993695a51f0ee5af3407dedb37">3d27c223a5</a></td><td>David M. Lee</td><td>Fixing extconf compile</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1a6f1a9f987f892b66f96786438e21f86bdef79">d1a6f1a9f9</a></td><td>Matt Jordan</td><td>git migration: Remove support for file versions</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a77c31b99c8e60507a268251ab8a941802793fca">a77c31b99c</a></td><td>Corey Farrell</td><td>main/editline: Add .gitignore.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d918c3b78eece62db33d4d37a45310437f41d0c4">d918c3b78e</a></td><td>Matt Jordan</td><td>.gitignore: Ignore tarballs (*.gz)</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=555b5f5d30b7fc061eb7ea7344e91cde93a73341">555b5f5d30</a></td><td>gtjoseph</td><td>Add .gitignore and .gitreview files</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5807ca519c5ec22249dd265749a983f54ea01953">5807ca519c</a></td><td>Matt Jordan</td><td>Blocked revisions 434708</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=16afee4651c034f5cab18b0ee0a1b925b9044fdf">16afee4651</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: Cleanup load unload</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=125acc52fea2dbace7ab8e4f68d6a09e975a0bc1">125acc52fe</a></td><td>Richard Mudgett</td><td>bridge_softmix.c,channel.c: Minor code simplification and cleanup.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c9791dba1fb77dffd9b8030a5b06aa16e195e381">c9791dba1f</a></td><td>Matt Jordan</td><td>res/ari: Fix model validation for ChannelHold event</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=73c286a393e359598073293d31eb9c4f07704af0">73c286a393</a></td><td>gtjoseph</td><td>loader/main: Don't set ast_fully_booted until deferred reloads are processed</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1695a5b85ffe22ae26c06b46543782a29b31df67">1695a5b85f</a></td><td>Richard Mudgett</td><td>chan_iax2.c: Fix ref leak in iax2_request().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=92c1688edb4652302de73dfba74f8e03d7dca62f">92c1688edb</a></td><td>Richard Mudgett</td><td>bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1712d16825876044f9c1ab072feb7aee09f6cf6b">1712d16825</a></td><td>Richard Mudgett</td><td>format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear().</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ae39dd1f4675036cadd7222a906f679aea4b58dd">ae39dd1f46</a></td><td>Matt Jordan</td><td>chan_iax2: Fix compilation issue due to funky merge</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a6aed7f6f661a7d65601804dd4bbf89c11e968e1">a6aed7f6f6</a></td><td>Scott Griepentrog</td><td>Revert accidental change in r434261</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0584e29300520211291fed42df6c17c90dce64c3">0584e29300</a></td><td>Scott Griepentrog</td><td>pjsip: resolve compatibility problem with ast_sip_session</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c516981dc7706979ae3e5d2a44927e005ec4d9f4">c516981dc7</a></td><td>Mark Michelson</td><td>Do not queue message requests that we do not respond to.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=169e57d2e066eb49ed2a881aabfbc212a46f01d2">169e57d2e0</a></td><td>Scott Griepentrog</td><td>pjsip: resolve compatibility problem with ast_sip_session</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1eb0c5f4e8a9095b312811f50c171d07d3f049e8">1eb0c5f4e8</a></td><td>Corey Farrell</td><td>Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e3011859833ce36c6879dcd2d601e7e1f363bcbe">e301185983</a></td><td>Ashley Sanders</td><td>stasis: set a channel variable on websocket disconnect error</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a1f12d9231b6b29f9f83d34e5bb2d64d5f203cfe">a1f12d9231</a></td><td>Ashley Sanders</td><td>stasis: set a channel variable on websocket disconnect error</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=94949e7f2f8cdd23b44b108348c5f3642964a367">94949e7f2f</a></td><td>Richard Mudgett</td><td>chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=99677396699ae415a8f58cf7436cb9c96d879f57">9967739669</a></td><td>Corey Farrell</td><td>Re-add _ast_mem_backtrace_buffer variable for ABI compatibility.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2d39bc55286c73cb81138f1dc8eaf8217ed312f1">2d39bc5528</a></td><td>Corey Farrell</td><td>Fix an ABI compatibility issue with ast_log_safe for modules.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cfbf5fbe918bc34f3d600760fc0b6f13a3a9a0ed">cfbf5fbe91</a></td><td>Jonathan Rose</td><td>SAC: Add a few basic queues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1a50d8d4c249b888bd93c256cec53fb7bcf8db1d">1a50d8d4c2</a></td><td>Jonathan Rose</td><td>SAC: Add conferencing extensions and configuration</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c6c08d755d429cbc43550bdc387fd3cd68eaa4d9">c6c08d755d</a></td><td>Rusty Newton</td><td>configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13557675d42a95557f4ee23f89dae6830cc01678">13557675d4</a></td><td>Richard Mudgett</td><td>res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dc2cf21144b81800329bad4596ded07ed0882ed7">dc2cf21144</a></td><td>Richard Mudgett</td><td>res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b0df413fb2756559aa261b5cf3374b8216f82054">b0df413fb2</a></td><td>Corey Farrell</td><td>Fix link error for utils/aelparse.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4b225e210485fa41b8cf00f6955a22101ff7ef6b">4b225e2104</a></td><td>Corey Farrell</td><td>Fix compile errors caused by r4500 / r4501.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dea885a6070a62dfcaa631740b86f7ebdc09c977">dea885a607</a></td><td>Richard Mudgett</td><td>A couple minor cleanup tweaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6ca98524bfc321c12398df5ed0cac6c7cad3c55c">6ca98524bf</a></td><td>Richard Mudgett</td><td>Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1c090281718bba6a5cee8f573f4dad0e46427496">1c09028171</a></td><td>Richard Mudgett</td><td>res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dba0f1ad678c5a058d86b9d896899a76f90c55b7">dba0f1ad67</a></td><td>Richard Mudgett</td><td>res_pjsip_session: Fix off-nominal extra unref of session.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2c7b9451499c5bd50bc6fa54a0073781b116834c">2c7b945149</a></td><td>Scott Griepentrog</td><td>Various: bugfixes found via chaos</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=1fb1c81923249413e3e439a645ade8793ccc0db6">1fb1c81923</a></td><td>Scott Griepentrog</td><td>core: Introduce chaos into memory allocations</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2122c205e677a0da93b582b148b6495b660e1e6d">2122c205e6</a></td><td>Richard Mudgett</td><td>Audit ast_sockaddr_resolve() usage for memory leaks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=522f0631863fb4ddf6337ecbdbf9b56c92eb757f">522f063186</a></td><td>Richard Mudgett</td><td>res_pjsip: Add reason comment.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34aa0214ebfc033d5deaa05da98bf9cb740188e3">34aa0214eb</a></td><td>Richard Mudgett</td><td>chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9fd61f2c7575cc32bc4a8411eac52195e995c97">b9fd61f2c7</a></td><td>Matt Jordan</td><td>main/audiohook: Update internal sample rate on reads</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bd029688cd5c2af9cae769fe31dfea74ef127318">bd029688cd</a></td><td>Richard Mudgett</td><td>res_pjsip: Move internal init/destroy prototypes to private header file.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=737064bfa4c6edb651c7d8d1efac780086cb9f6b">737064bfa4</a></td><td>Richard Mudgett</td><td>res_pjsip: Fixed invalid empty Server and User-Agent SIP headers.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bc357c1d7efd2fc7fce8c21fd472f437a9f6a7df">bc357c1d7e</a></td><td>Joshua Colp</td><td>core: Don't create snapshots with locks.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e158517a9c39ad22d2959f010d905ad2bc9872d4">e158517a9c</a></td><td>Richard Mudgett</td><td>res_pjsip_refer: Make safely get the context for a blind transfer.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5d16d80b59a797764a5d40b685ae79d4116d0539">5d16d80b59</a></td><td>Richard Mudgett</td><td>res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=772793f18eba63a5f6426fb18e51bf6defba1032">772793f18e</a></td><td>Jonathan Rose</td><td>app: Add functions to swap voicemail function table for testing purposes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8cced7767c09c7300f47c2cc445227f0f488b899">8cced7767c</a></td><td>Richard Mudgett</td><td>chan_dahdi/sig_analog: Fix distinctive ring detection to suck less.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13e715b30c1e4fff81d6c42896a5ef5ade29b74a">13e715b30c</a></td><td>Richard Mudgett</td><td>chan_sip: Fix realtime locking inversion when poking a just built peer.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=06fa8db8646faa84e0506235623d7d33a0bf6f82">06fa8db864</a></td><td>gtjoseph</td><td>app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=999d96d40525ca6bacb53fb09b1686fa260dd23d">999d96d405</a></td><td>Matt Jordan</td><td>translate: Prevent invalid memory accesses on fast shutdown</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c33c5183a51cf1d1343444108a09614538cd1ee2">c33c5183a5</a></td><td>Scott Griepentrog</td><td>Dial API: add self destruct option when complete</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=53aec7a969cc9f99fafe3ceeb912ec6b5d50bdcb">53aec7a969</a></td><td>Rusty Newton</td><td>configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=474fec4f92eb71e0bfa27cacac3281690b8ccf32">474fec4f92</a></td><td>Matt Jordan</td><td>configure: Promote SQLite3 "not installed" warning to error</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43a3e80be155dc7569a94082a1d1c1484cf40dbe">43a3e80be1</a></td><td>David M. Lee</td><td>Increase WebSocket frame size and improve large read handling</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89b48af3e572a25ede413ed89b6d765969fa5bec">89b48af3e5</a></td><td>Richard Mudgett</td><td>chan_dahdi/sig_analog: Put log message strings on one line.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e0ff83c27276f51b20faf081ed85e47f3c424384">e0ff83c272</a></td><td>Richard Mudgett</td><td>chan_dahdi: Remove some dead code.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2181c9443f4c43d0c463151f0930b8a8af888c1d">2181c9443f</a></td><td>Richard Mudgett</td><td>res_pjsip_refer: Handle INVITE with Replaces failure after answer.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c8f3074cc4b96a884d67c70b74dfacf68272773a">c8f3074cc4</a></td><td>Joshua Colp</td><td>res_sorcery_config: Improve object lookup times.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4f4d03fdd12427a70accb6557efef7ef40cac3b0">4f4d03fdd1</a></td><td>Matt Jordan</td><td>apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3543a363624261f61d7b985a6f5bb2d700f2ab64">3543a36362</a></td><td>Joshua Colp</td><td>'information' ends with an 'n'.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4d8ab20a8abac76c680a40605bb6e6e8e675afde">4d8ab20a8a</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: Add ability to auto-create hints.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=323c0927ac63ce1c13133961bd46abbf6d265a18">323c0927ac</a></td><td>Scott Griepentrog</td><td>various: cleanup issues found during leak hunt</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=18c8c1bae39e4a236f4cc5b2cfc1d759af624593">18c8c1bae3</a></td><td>Joshua Colp</td><td>res_pjsip_keepalive: Don't crash if PJSIP module is not loaded.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e8896ac00845850e9d88dadcd1d6dc30a476c1fc">e8896ac008</a></td><td>Mark Michelson</td><td>Use SIPS URIs in Contact headers when appropriate.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b8ea23b0d12882065c448a9fb8ce1de503dc2103">b8ea23b0d1</a></td><td>Mark Michelson</td><td>Allow disabling of 100rel support on PJSIP endpoints.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6e5eb9af88d25f8743c9aeb724ce979b60a6e6c0">6e5eb9af88</a></td><td>gtjoseph</td><td>res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c3add776af2fc339de0c07f336d8b6eb9d55d062">c3add776af</a></td><td>Sean Bright</td><td>media formats: update res_format_attr_opus &amp; silk</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b64f4bb6ee54044f44ee1322f2b2c89ebcd968d1">b64f4bb6ee</a></td><td>Joshua Colp</td><td>bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7f9b28b0c6d53837049b026103d56467b3f89e7d">7f9b28b0c6</a></td><td>Matt Jordan</td><td>ARI: Improve wiki documentation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ceedd4037029fe8d61cccbdcbbd42ee1993c5e6f">ceedd40370</a></td><td>Joshua Colp</td><td>res_parking: Fix crash due to race condition when unloading.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e302116e40811660c39e2ceab54d1120e19d5990">e302116e40</a></td><td>Richard Mudgett</td><td>app_confbridge: Make CBRec channel names more unique.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=f8b3fb6e2fce6f01a01c82d4297584e574c88aec">f8b3fb6e2f</a></td><td>Richard Mudgett</td><td>app_confbridge: Whitespace</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=197265438eb8583730d0ff7a0629731db47a3dc1">197265438e</a></td><td>David M. Lee</td><td>Add depend on pjproject to res_pjsip_config_wizard.c</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e23f07beb80a76160edec656d6509db5f1a2885a">e23f07beb8</a></td><td>Walter Doekes</td><td>Fix typo's (retrieve, specified, address).</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c73b4b2a46692be957d0d7ef193f75dc5c98be59">c73b4b2a46</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Minor code cleanup.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5e10007dbde4dcad25776708000ddd027d678cf7">5e10007dbd</a></td><td>Richard Mudgett</td><td>res_pjsip_outbound_registration.c: Move unref to a better place.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=98c3983c893149420786070356c51e406504a617">98c3983c89</a></td><td>Matt Jordan</td><td>main/rtp_engine: Format NTP timestamps as unsigned longs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ab5af1f3d8180cdba1b638bdec603ef5ee893ddd">ab5af1f3d8</a></td><td>Mark Michelson</td><td>Call extension state callbacks at hint creation.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=34c220203f38623689a5329d8aa001c552cc1386">34c220203f</a></td><td>Kevin Harwell</td><td>REVERTING res_pjsip: make it unloadable</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e257244bbb8014ab97ea98a231ce2f2e21d1228f">e257244bbb</a></td><td>Mark Michelson</td><td>Change PJProject version requirement for ca_list_path transport option in CHANGES file.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fa80d9658df8aff71a8975ab7d1fe477ea3f99b9">fa80d9658d</a></td><td>Richard Mudgett</td><td>res_fax.c, res_fax_spandsp.c: Remove redundant locking.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6c426e86bd470c3a2f3b73b3a070583c92c19af4">6c426e86bd</a></td><td>Richard Mudgett</td><td>res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function definitions.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=c95391f23ccc30e7d495a16ef927846dab96984a">c95391f23c</a></td><td>Joshua Colp</td><td>res_pjsip_outbound_registration: Fix race condition when reloading and listing registrations.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=eb9ce791d8e59d2aa63fe13518436f46b61b4a44">eb9ce791d8</a></td><td>Kinsey Moore</td><td>res_fax: Add T.38 negotiation timeout option</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b937438c17dfcd0696931fb3abde9ae816578c32">b937438c17</a></td><td>gtjoseph</td><td>res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=143bec54ee134c9fcd014c3445ea64af4793f5b0">143bec54ee</a></td><td>gtjoseph</td><td>res_pjsip_outbound_registration: Fix reference leak.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=6e59bf649112ed371311081f91772f053eac3e94">6e59bf6491</a></td><td>gtjoseph</td><td>res_pjsip_outbound_registration: Fix several reload issues</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a10d2966b67a2e52811a7e4fa636b630f1fef741">a10d2966b6</a></td><td>gtjoseph</td><td>res_pjsip_exten_state: Change 'does not exist' warning to notice</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=13ed8f73edbe95c0455a09d7dcc05d7da8aa4643">13ed8f73ed</a></td><td>gtjoseph</td><td>res_pjsip_mwi: Change "MWI Subscription failed" message from warning to notice</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=42e4cb7174225c33cbc18479eb4b99fa6896d478">42e4cb7174</a></td><td>gtjoseph</td><td>func_config: Add ability to retrieve specific occurrence of a variable</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=75cd302b0af6a14d8a7b53330049b34e33aece75">75cd302b0a</a></td><td>gtjoseph</td><td>config: Add option to NOT preserve effective context when changing a template</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e17a1a8ba1a250a2a4850af1f451d3f03995cace">e17a1a8ba1</a></td><td>Kinsey Moore</td><td>Fix dev-mode build on recent gcc</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=dd42e92e7ab948459129cc8933ff595681589dba">dd42e92e7a</a></td><td>Matt Jordan</td><td>contrib/ast-db-manage: Correct down_revision path for user_eq_phone</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4becfae3b1912828a9f3f2273f33da65c09b45f7">4becfae3b1</a></td><td>gtjoseph</td><td>res_pjsip_mwi: Change warning to notice</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=9d457fe5c297125c709de77eb115c99127aea508">9d457fe5c2</a></td><td>gtjoseph</td><td>bridge_native_rtp: Change local/remote message from debug/2 to verb/4</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0fa6c34dc6fb6927baf58d19cff326577745c12c">0fa6c34dc6</a></td><td>gtjoseph</td><td>outbound_registration: Add 'pjsip send register' and update 'send unregister'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d873b09075bbcc584b936ec1f46e0f58c7ec7279">d873b09075</a></td><td>gtjoseph</td><td>pjsip cli: Fix sorting of contacts for 'pjsip list contacts'</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b9a7875dd6a14a13981e5141b4464c21263b6346">b9a7875dd6</a></td><td>Joshua Colp</td><td>pjsip: Document addition of 'PJSIP_AOR' and 'PJSIP_CONTACT' in CHANGES file.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=cca262e7d32944ae5b0b6f8e0de080be1149bf74">cca262e7d3</a></td><td>Kinsey Moore</td><td>PJSIP: Update transport method documentation</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=d1c532034b6f281dfb2cda8a3294da0d725001e6">d1c532034b</a></td><td>gtjoseph</td><td>pjsip_options: Fix continued qualifies after endpoint/aor deletion</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0a3dd7589ebf6845a95b561a08ffb239b5ed2b07">0a3dd7589e</a></td><td>gtjoseph</td><td>test_astobj2: Fix warning for missing trailing slash in category</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fca0be57d9207ba9ceef650cf79802f031f036d3">fca0be57d9</a></td><td>Richard Mudgett</td><td>queue_log: Post QUEUESTART entry when Asterisk fully boots.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=fc79cf6428213191023eaed6ab25e89dbfde5564">fc79cf6428</a></td><td>gtjoseph</td><td>res_pjsip_phoneprovi_provider: Fix reload</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7074bf956b1ec215b55176c5f0137924450a2749">7074bf956b</a></td><td>Richard Mudgett</td><td>chan_dahdi: Don't ignore setvar when using configuration section scheme.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=e603fbe04a678e978562ca642e4369c820601c29">e603fbe04a</a></td><td>Richard Mudgett</td><td>chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=14d2f8f20fc71b3aed7191c69b9fda378b696840">14d2f8f20f</a></td><td>Mark Michelson</td><td>Prevent potential infinite outbound authentication loops in registration.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=5bd5f580c13e5d96375a3aeeba1c2a0d3eeac17c">5bd5f580c1</a></td><td>Mark Michelson</td><td>Ensure the correct value is returned for CHANNEL(pjsip, secure)</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b4621cd0f538dc71a0418b57750a79e39ace2944">b4621cd0f5</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: fix unload SEGV</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=105f224cfd9a3651cb78f21e4f60b656013be488">105f224cfd</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: Change FILEUNCHANGED config_load2 flag determination</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a3534b7c0547db14c87f3f2622786a40374a74ee">a3534b7c05</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: fix test breakage</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=ad85e54fd9b545aa6c1dd43a14bdf92344cf4fa0">ad85e54fd9</a></td><td>Joshua Colp</td><td>res_pjsip_t38: Fix T.38 failure when peer reinvites immediately.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=89617370ec0e0cd0384ee24c451d2c70b51ec222">89617370ec</a></td><td>gtjoseph</td><td>res_pjsip_config_wizard: Allow streamlined config of common pjsip scenarios</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b85f79c0c1aa8f77492a09b91a077daef1192ff8">b85f79c0c1</a></td><td>Mark Michelson</td><td>Activate persistent subscriptions when they are recreated.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2b8c4410964ef037450626099c65636340190312">2b8c441096</a></td><td>gtjoseph</td><td>loader: Move definition of ast_module_reload from _private.h to module.h</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=8c019b1a6b710d67fd4ad55f7f74aa99e8674fac">8c019b1a6b</a></td><td>Matt Jordan</td><td>res/res_agi: Make Verbose message for 'stream file' match other playbacks</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=7ff0d266a66759307885787d3ef5e7209c080510">7ff0d266a6</a></td><td>Matt Jordan</td><td>Add 11 merge properties</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49386cf5680d2348d3511e2a68d41bd1cdfdd4d4">49386cf568</a></td><td>David M. Lee</td><td>Fix crash for sorcery misconfigs</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3b0c40f337deb199dab49f91ea1c73f805df0cb1">3b0c40f337</a></td><td>Kinsey Moore</td><td>PJSIP: Allow use of 'inactive' streams for hold</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=15af40180abd6aea2df32241bc14b77a3bda5c32">15af40180a</a></td><td>Kinsey Moore</td><td>Sorcery: Log when old config remains in use</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=0c9fbb449f272b826e022499dddda39bc12dda7e">0c9fbb449f</a></td><td>Joshua Colp</td><td>res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=2288f910ea66ee9a744f5ee5ee63414faa50712a">2288f910ea</a></td><td>Kinsey Moore</td><td></td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=b7f7d045ac9dc82b09eec377dc7e879ba938a128">b7f7d045ac</a></td><td>Kinsey Moore</td><td>language key into account.</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=50f651729683529139e8edcb0a71342ecc905605">50f6517296</a></td><td>Kinsey Moore</td><td>Stasis: Update unittest for channel snapshots</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a220a08777a3d00595cdc65edafee6a837983aca">a220a08777</a></td><td>Kinsey Moore</td><td>PJSIP: Fix assert on initial mass qualify</td></tr>
<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=22a91bf69839f5a5191342c1e508cb8fd7d86d50">22a91bf698</a></td><td>Scott Griepentrog</td><td>core: avoid possible asterisk -r crash from long id</td></tr>
</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-certified-13.1-cert4-summary.html | 30
asterisk-certified-13.1-cert4-summary.txt | 127
b/.gitignore | 6
b/.version | 2
b/CHANGES | 412
b/ChangeLog |17975 ++++++++++
b/Makefile | 140
b/Makefile.moddir_rules | 5
b/Makefile.rules | 42
b/UPGRADE.txt | 92
b/addons/chan_mobile.c | 2
b/addons/chan_ooh323.c | 15
b/addons/ooh323c/src/ooh245.c | 2
b/addons/ooh323c/src/ooq931.c | 6
b/addons/ooh323c/src/printHandler.c | 2
b/addons/res_config_mysql.c | 72
b/apps/Makefile | 4
b/apps/app_adsiprog.c | 2
b/apps/app_agent_pool.c | 9
b/apps/app_alarmreceiver.c | 2
b/apps/app_amd.c | 19
b/apps/app_chanisavail.c | 7
b/apps/app_chanspy.c | 8
b/apps/app_confbridge.c | 344
b/apps/app_dial.c | 29
b/apps/app_dictate.c | 4
b/apps/app_dumpchan.c | 2
b/apps/app_fax.c | 4
b/apps/app_getcpeid.c | 10
b/apps/app_macro.c | 31
b/apps/app_meetme.c | 40
b/apps/app_minivm.c | 24
b/apps/app_mixmonitor.c | 15
b/apps/app_osplookup.c | 4
b/apps/app_page.c | 2
b/apps/app_playback.c | 4
b/apps/app_queue.c | 528
b/apps/app_sms.c | 14
b/apps/app_stasis.c | 12
b/apps/app_voicemail.c | 151
b/apps/confbridge/conf_chan_record.c | 7
b/apps/confbridge/conf_config_parser.c | 32
b/apps/confbridge/conf_state_multi_marked.c | 12
b/autoconf/ast_check_raii.m4 | 56
b/autoconf/ast_check_strsep_array_bounds.m4 | 81
b/autoconf/ast_gcc_attribute.m4 | 2
b/bridges/bridge_builtin_features.c | 8
b/bridges/bridge_holding.c | 20
b/bridges/bridge_native_rtp.c | 31
b/bridges/bridge_simple.c | 14
b/bridges/bridge_softmix.c | 191
b/build_tools/cflags.xml | 36
b/build_tools/get_moduleinfo | 5
b/build_tools/make_buildopts_h | 32
b/build_tools/make_check_alembic | 29
b/build_tools/make_version_c | 25
b/build_tools/menuselect-deps.in | 4
b/build_tools/mkpkgconfig | 9
b/cdr/cdr_csv.c | 10
b/cdr/cdr_manager.c | 126
b/cdr/cdr_odbc.c | 32
b/cdr/cdr_pgsql.c | 44
b/cel/cel_manager.c | 160
b/cel/cel_pgsql.c | 4
b/cel/cel_radius.c | 4
b/cel/cel_sqlite3_custom.c | 1
b/channels/Makefile | 10
b/channels/chan_alsa.c | 2
b/channels/chan_console.c | 2
b/channels/chan_dahdi.c | 183
b/channels/chan_dahdi.h | 2
b/channels/chan_iax2.c | 385
b/channels/chan_mgcp.c | 28
b/channels/chan_misdn.c | 10
b/channels/chan_motif.c | 3
b/channels/chan_nbs.c | 2
b/channels/chan_oss.c | 2
b/channels/chan_phone.c | 2
b/channels/chan_pjsip.c | 312
b/channels/chan_sip.c | 2316 -
b/channels/chan_skinny.c | 66
b/channels/chan_unistim.c | 66
b/channels/chan_vpb.cc | 7
b/channels/dahdi/bridge_native_dahdi.c | 15
b/channels/iax2/parser.c | 2
b/channels/misdn/Makefile | 2
b/channels/misdn/ie.c | 14
b/channels/misdn_config.c | 2
b/channels/pjsip/dialplan_functions.c | 55
b/channels/sig_analog.c | 205
b/channels/sig_analog.h | 1
b/channels/sig_pri.c | 125
b/channels/sig_pri.h | 2
b/channels/sip/dialplan_functions.c | 4
b/channels/sip/include/dialog.h | 41
b/channels/sip/include/route.h | 2
b/channels/sip/include/sip.h | 20
b/channels/sip/reqresp_parser.c | 2
b/channels/vcodecs.c | 6
b/codecs/codec_gsm.c | 29
b/codecs/codec_ilbc.c | 28
b/codecs/codec_lpc10.c | 41
b/codecs/codec_resample.c | 8
b/codecs/codec_speex.c | 60
b/codecs/gsm/Makefile | 2
b/codecs/gsm/src/gsm_create.c | 2
b/configs/basic-pbx/README | 15
b/configs/basic-pbx/asterisk.conf | 26
b/configs/basic-pbx/cdr.conf | 7
b/configs/basic-pbx/cdr_custom.conf | 4
b/configs/basic-pbx/confbridge.conf | 1
b/configs/basic-pbx/extensions.conf | 193
b/configs/basic-pbx/indications.conf | 19
b/configs/basic-pbx/logger.conf | 9
b/configs/basic-pbx/modules.conf | 116
b/configs/basic-pbx/musiconhold.conf | 5
b/configs/basic-pbx/pjsip.conf | 332
b/configs/basic-pbx/queues.conf | 19
b/configs/basic-pbx/voicemail.conf | 23
b/configs/samples/amd.conf.sample | 29
b/configs/samples/cdr.conf.sample | 2
b/configs/samples/cdr_odbc.conf.sample | 1
b/configs/samples/chan_dahdi.conf.sample | 10
b/configs/samples/features.conf.sample | 7
b/configs/samples/http.conf.sample | 10
b/configs/samples/iax.conf.sample | 7
b/configs/samples/pjproject.conf.sample | 28
b/configs/samples/pjsip.conf.sample | 59
b/configs/samples/pjsip_wizard.conf.sample | 147
b/configs/samples/queues.conf.sample | 14
b/configs/samples/res_fax.conf.sample | 4
b/configs/samples/rtp.conf.sample | 27
b/configs/samples/sip.conf.sample | 10
b/configure | 1631
b/configure.ac | 201
b/contrib/ast-db-manage/config/env.py | 3
b/contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py | 10
b/contrib/ast-db-manage/config/versions/136885b81223_add_regcontext_to_pj.py | 21
b/contrib/ast-db-manage/config/versions/154177371065_add_default_from_user.py | 7
b/contrib/ast-db-manage/config/versions/1758e8bbf6b_increase_useragent_column_size.py | 6
b/contrib/ast-db-manage/config/versions/189a235b3fd7_add_keep_alive_interval.py | 23
b/contrib/ast-db-manage/config/versions/1d50859ed02e_create_accountcode.py | 3
b/contrib/ast-db-manage/config/versions/21e526ad3040_add_pjsip_debug_option.py | 3
b/contrib/ast-db-manage/config/versions/23530d604b96_add_rpid_immediate.py | 49
b/contrib/ast-db-manage/config/versions/26d7f3bf0fa5_add_bind_rtp_to_media_address_to_pjsip.py | 32
b/contrib/ast-db-manage/config/versions/26f10cadc157_add_pjsip_timeout_options.py | 25
b/contrib/ast-db-manage/config/versions/28b8e71e541f_add_g726_non_standard.py | 31
b/contrib/ast-db-manage/config/versions/28ce1e718f05_add_fatal_response_interval.py | 3
b/contrib/ast-db-manage/config/versions/2d078ec071b7_increaes_contact_column_size.py | 24
b/contrib/ast-db-manage/config/versions/2fc7930b41b3_add_pjsip_endpoint_options_for_12_1.py | 31
b/contrib/ast-db-manage/config/versions/31cd4f4891ec_add_auto_dtmf_mode.py | 64
b/contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py | 3
b/contrib/ast-db-manage/config/versions/3855ee4e5f85_add_missing_pjsip_options.py | 6
b/contrib/ast-db-manage/config/versions/3bcc0b5bc2c9_add_allow_reload_to_ps_transports.py | 26
b/contrib/ast-db-manage/config/versions/423f34ad36e2_fix_pjsip_qualify_ti.py | 26
b/contrib/ast-db-manage/config/versions/43956d550a44_add_tables_for_pjsip.py | 13
b/contrib/ast-db-manage/config/versions/45e3f47c6c44_add_pjsip_endpoint_identifier_order.py | 3
b/contrib/ast-db-manage/config/versions/461d7d691209_add_pjsip_qualify_timeout.py | 24
b/contrib/ast-db-manage/config/versions/498357a710ae_add_rtp_keepalive.py | 5
b/contrib/ast-db-manage/config/versions/4c573e7135bd_fix_tos_field_types.py | 54
b/contrib/ast-db-manage/config/versions/4da0c5f79a9c_create_tables.py | 14
b/contrib/ast-db-manage/config/versions/5139253c0423_make_q_member_uniqueid_autoinc.py | 33
b/contrib/ast-db-manage/config/versions/51f8cb66540e_add_further_dtls_options.py | 5
b/contrib/ast-db-manage/config/versions/5950038a6ead_fix_pjsip_verifiy_typo.py | 6
b/contrib/ast-db-manage/config/versions/a541e0b5e89_add_pjsip_max_initial_qualify_time.py | 21
b/contrib/ast-db-manage/config/versions/dbc44d5a908_add_missing_columns_to_sys_and_reg.py | 36
b/contrib/ast-db-manage/config/versions/e96a0b8071c_increase_pjsip_column_size.py | 33
b/contrib/ast-db-manage/config/versions/eb88a14f2a_add_media_encryption_optimistic_to_pjsip.py | 3
b/contrib/docker/Dockerfile.asterisk | 19
b/contrib/docker/Dockerfile.packager | 9
b/contrib/docker/README.md | 39
b/contrib/docker/make-package.sh | 72
b/contrib/editors/asterisk.vim | 4
b/contrib/realtime/mssql/mssql_cdr.sql | 4
b/contrib/realtime/mssql/mssql_config.sql | 284
b/contrib/realtime/mssql/mssql_voicemail.sql | 10
b/contrib/realtime/mysql/mysql_cdr.sql | 2
b/contrib/realtime/mysql/mysql_config.sql | 164
b/contrib/realtime/mysql/mysql_voicemail.sql | 6
b/contrib/realtime/oracle/oracle_cdr.sql | 10
b/contrib/realtime/oracle/oracle_config.sql | 288
b/contrib/realtime/oracle/oracle_voicemail.sql | 16
b/contrib/realtime/postgresql/postgresql_cdr.sql | 2
b/contrib/realtime/postgresql/postgresql_config.sql | 208
b/contrib/realtime/postgresql/postgresql_voicemail.sql | 6
b/contrib/scripts/astversion | 536
b/contrib/scripts/autosupport | 12
b/contrib/scripts/clang-scan-build | 136
b/contrib/scripts/install_prereq | 18
b/contrib/scripts/sip_to_pjsip/astconfigparser.py | 15
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 23
b/contrib/utils/eagi_proxy.c | 2
b/doc/.gitignore | 3
b/doc/asterisk-ng-doxygen.in | 1688
b/formats/format_wav.c | 53
b/formats/format_wav_gsm.c | 16
b/funcs/func_callerid.c | 60
b/funcs/func_cdr.c | 50
b/funcs/func_channel.c | 70
b/funcs/func_config.c | 46
b/funcs/func_curl.c | 5
b/funcs/func_env.c | 4
b/funcs/func_groupcount.c | 2
b/funcs/func_hangupcause.c | 2
b/funcs/func_holdintercept.c | 236
b/funcs/func_iconv.c | 3
b/funcs/func_math.c | 12
b/funcs/func_odbc.c | 1
b/funcs/func_periodic_hook.c | 10
b/funcs/func_pjsip_aor.c | 186
b/funcs/func_pjsip_contact.c | 203
b/funcs/func_presencestate.c | 16
b/include/asterisk.h | 20
b/include/asterisk/_private.h | 35
b/include/asterisk/app.h | 4
b/include/asterisk/ari.h | 5
b/include/asterisk/ast_version.h | 3
b/include/asterisk/audiohook.h | 1
b/include/asterisk/autochan.h | 20
b/include/asterisk/autoconfig.h.in | 40
b/include/asterisk/bridge.h | 14
b/include/asterisk/bridge_channel.h | 21
b/include/asterisk/bridge_channel_internal.h | 82
b/include/asterisk/bridge_internal.h | 3
b/include/asterisk/bridge_technology.h | 5
b/include/asterisk/cel.h | 4
b/include/asterisk/channel.h | 83
b/include/asterisk/config.h | 66
b/include/asterisk/core_local.h | 32
b/include/asterisk/dial.h | 1
b/include/asterisk/dsp.h | 3
b/include/asterisk/endpoints.h | 10
b/include/asterisk/format.h | 23
b/include/asterisk/format_cap.h | 5
b/include/asterisk/http.h | 22
b/include/asterisk/http_websocket.h | 90
b/include/asterisk/inline_api.h | 12
b/include/asterisk/json.h | 18
b/include/asterisk/lock.h | 2
b/include/asterisk/logger.h | 57
b/include/asterisk/manager.h | 53
b/include/asterisk/module.h | 94
b/include/asterisk/monitor.h | 20
b/include/asterisk/pbx.h | 16
b/include/asterisk/res_fax.h | 6
b/include/asterisk/res_odbc.h | 78
b/include/asterisk/res_odbc_transaction.h | 54
b/include/asterisk/res_pjproject.h | 96
b/include/asterisk/res_pjsip.h | 429
b/include/asterisk/res_pjsip_cli.h | 2
b/include/asterisk/res_pjsip_session.h | 18
b/include/asterisk/rtp_engine.h | 24
b/include/asterisk/sched.h | 11
b/include/asterisk/select.h | 4
b/include/asterisk/sem.h | 19
b/include/asterisk/sip_api.h | 1
b/include/asterisk/slin.h | 4
b/include/asterisk/sorcery.h | 130
b/include/asterisk/stasis.h | 11
b/include/asterisk/stasis_app.h | 26
b/include/asterisk/stasis_cache_pattern.h | 19
b/include/asterisk/stasis_endpoints.h | 6
b/include/asterisk/statsd.h | 71
b/include/asterisk/strings.h | 19
b/include/asterisk/syslog.h | 2
b/include/asterisk/taskprocessor.h | 27
b/include/asterisk/term.h | 4
b/include/asterisk/test.h | 35
b/include/asterisk/threadpool.h | 53
b/include/asterisk/time.h | 10
b/include/asterisk/translate.h | 8
b/include/asterisk/utils.h | 112
b/include/asterisk/vector.h | 504
b/main/.gitignore | 3
b/main/Makefile | 126
b/main/aoc.c | 108
b/main/app.c | 11
b/main/asterisk.c | 278
b/main/astfd.c | 57
b/main/astmm.c | 107
b/main/astobj2.c | 14
b/main/astobj2_container.c | 6
b/main/astobj2_hash.c | 13
b/main/astobj2_rbtree.c | 2
b/main/audiohook.c | 205
b/main/autochan.c | 6
b/main/bridge.c | 237
b/main/bridge_basic.c | 28
b/main/bridge_channel.c | 228
b/main/bucket.c | 7
b/main/callerid.c | 13
b/main/ccss.c | 4
b/main/cdr.c | 32
b/main/cel.c | 18
b/main/channel.c | 456
b/main/channel_internal_api.c | 49
b/main/cli.c | 30
b/main/codec.c | 4
b/main/codec_builtin.c | 4
b/main/config.c | 74
b/main/config_options.c | 6
b/main/core_local.c | 39
b/main/db.c | 13
b/main/devicestate.c | 25
b/main/dial.c | 8
b/main/dsp.c | 34
b/main/editline/np/strlcat.c | 8
b/main/editline/np/strlcpy.c | 10
b/main/endpoints.c | 39
b/main/enum.c | 2
b/main/event.c | 2
b/main/features.c | 8
b/main/file.c | 6
b/main/format.c | 38
b/main/format_cache.c | 1
b/main/format_cap.c | 43
b/main/framehook.c | 2
b/main/hashtab.c | 6
b/main/http.c | 180
b/main/indications.c | 5
b/main/jitterbuf.c | 2
b/main/json.c | 29
b/main/libasteriskpj.c | 52
b/main/libasteriskssl.c | 35
b/main/loader.c | 270
b/main/logger.c | 362
b/main/manager.c | 140
b/main/manager_bridges.c | 68
b/main/manager_channels.c | 17
b/main/manager_endpoints.c | 1
b/main/message.c | 2
b/main/named_acl.c | 5
b/main/pbx.c |12879 ++-----
b/main/pbx_app.c | 510
b/main/pbx_builtins.c | 1438
b/main/pbx_functions.c | 723
b/main/pbx_hangup_handler.c | 300
b/main/pbx_private.h | 46
b/main/pbx_switch.c | 133
b/main/pbx_timing.c | 294
b/main/pbx_variables.c | 1180
b/main/presencestate.c | 2
b/main/rtp_engine.c | 214
b/main/sched.c | 88
b/main/sdp_srtp.c | 110
b/main/security_events.c | 7
b/main/sem.c | 33
b/main/sorcery.c | 412
b/main/stasis.c | 27
b/main/stasis_bridges.c | 3
b/main/stasis_cache_pattern.c | 34
b/main/stasis_channels.c | 79
b/main/stasis_endpoints.c | 143
b/main/stasis_message_router.c | 4
b/main/stdtime/localtime.c | 293
b/main/strings.c | 97
b/main/syslog.c | 8
b/main/taskprocessor.c | 168
b/main/tcptls.c | 9
b/main/term.c | 28
b/main/test.c | 80
b/main/threadpool.c | 139
b/main/translate.c | 156
b/main/udptl.c | 11
b/main/utils.c | 74
b/main/uuid.c | 2
b/main/xmldoc.c | 171
b/makeopts.in | 6
b/menuselect/configure | 8
b/menuselect/menuselect.c | 12
b/menuselect/menuselect.h | 2
b/menuselect/menuselect_curses.c | 73
b/pbx/Makefile | 2
b/pbx/dundi-parser.c | 8
b/pbx/pbx_config.c | 28
b/pbx/pbx_dundi.c | 21
b/pbx/pbx_spool.c | 35
b/res/Makefile | 12
b/res/ari.make | 11
b/res/ari/ari_model_validators.c | 862
b/res/ari/ari_model_validators.h | 210
b/res/ari/ari_websockets.c | 9
b/res/ari/config.c | 72
b/res/ari/resource_asterisk.c | 614
b/res/ari/resource_asterisk.h | 190
b/res/ari/resource_bridges.c | 22
b/res/ari/resource_bridges.h | 12
b/res/ari/resource_channels.c | 220
b/res/ari/resource_channels.h | 54
b/res/ari/resource_device_states.c | 5
b/res/ari/resource_endpoints.c | 61
b/res/ari/resource_events.c | 61
b/res/ari/resource_events.h | 17
b/res/ari/resource_mailboxes.c | 5
b/res/ari/resource_playbacks.c | 5
b/res/ari/resource_recordings.c | 5
b/res/parking/parking_applications.c | 9
b/res/parking/parking_manager.c | 92
b/res/parking/parking_tests.c | 9
b/res/res_agi.c | 6
b/res/res_ari.c | 26
b/res/res_ari_applications.c | 1
b/res/res_ari_asterisk.c | 1041
b/res/res_ari_bridges.c | 15
b/res/res_ari_channels.c | 137
b/res/res_ari_device_states.c | 1
b/res/res_ari_endpoints.c | 2
b/res/res_ari_events.c | 115
b/res/res_ari_mailboxes.c | 1
b/res/res_ari_playbacks.c | 1
b/res/res_ari_recordings.c | 1
b/res/res_ari_sounds.c | 1
b/res/res_calendar.c | 12
b/res/res_chan_stats.c | 4
b/res/res_config_odbc.c | 8
b/res/res_config_pgsql.c | 8
b/res/res_config_sqlite.c | 8
b/res/res_config_sqlite3.c | 16
b/res/res_crypto.c | 10
b/res/res_endpoint_stats.c | 157
b/res/res_fax.c | 146
b/res/res_fax_spandsp.c | 22
b/res/res_format_attr_h264.c | 20
b/res/res_format_attr_opus.c | 220
b/res/res_format_attr_silk.c | 26
b/res/res_format_attr_vp8.c | 228
b/res/res_hep_rtcp.c | 2
b/res/res_http_websocket.c | 241
b/res/res_manager_devicestate.c | 8
b/res/res_manager_presencestate.c | 8
b/res/res_monitor.c | 1
b/res/res_musiconhold.c | 5
b/res/res_mwi_external.c | 13
b/res/res_mwi_external_ami.c | 14
b/res/res_odbc.c | 1200
b/res/res_odbc.exports.in | 17
b/res/res_odbc_transaction.c | 529
b/res/res_odbc_transaction.exports.in | 6
b/res/res_phoneprov.c | 22
b/res/res_pjproject.c | 458
b/res/res_pjproject.exports.in | 6
b/res/res_pjsip.c | 808
b/res/res_pjsip/config_auth.c | 18
b/res/res_pjsip/config_domain_aliases.c | 1
b/res/res_pjsip/config_global.c | 199
b/res/res_pjsip/config_system.c | 35
b/res/res_pjsip/config_transport.c | 846
b/res/res_pjsip/include/res_pjsip_private.h | 210
b/res/res_pjsip/location.c | 284
b/res/res_pjsip/pjsip_cli.c | 40
b/res/res_pjsip/pjsip_configuration.c | 320
b/res/res_pjsip/pjsip_distributor.c | 17
b/res/res_pjsip/pjsip_global_headers.c | 19
b/res/res_pjsip/pjsip_options.c | 279
b/res/res_pjsip/pjsip_outbound_auth.c | 9
b/res/res_pjsip/presence_xml.c | 12
b/res/res_pjsip_acl.c | 1
b/res/res_pjsip_caller_id.c | 164
b/res/res_pjsip_config_wizard.c | 1307
b/res/res_pjsip_diversion.c | 103
b/res/res_pjsip_dlg_options.c | 2
b/res/res_pjsip_dtmf_info.c | 12
b/res/res_pjsip_endpoint_identifier_anonymous.c | 20
b/res/res_pjsip_endpoint_identifier_ip.c | 64
b/res/res_pjsip_endpoint_identifier_user.c | 22
b/res/res_pjsip_exten_state.c | 11
b/res/res_pjsip_history.c | 1353
b/res/res_pjsip_keepalive.c | 4
b/res/res_pjsip_messaging.c | 8
b/res/res_pjsip_multihomed.c | 25
b/res/res_pjsip_mwi.c | 118
b/res/res_pjsip_nat.c | 42
b/res/res_pjsip_notify.c | 5
b/res/res_pjsip_outbound_authenticator_digest.c | 10
b/res/res_pjsip_outbound_publish.c | 58
b/res/res_pjsip_outbound_registration.c | 707
b/res/res_pjsip_path.c | 6
b/res/res_pjsip_phoneprov_provider.c | 159
b/res/res_pjsip_publish_asterisk.c | 1
b/res/res_pjsip_pubsub.c | 119
b/res/res_pjsip_refer.c | 314
b/res/res_pjsip_registrar.c | 30
b/res/res_pjsip_registrar_expire.c | 100
b/res/res_pjsip_rfc3326.c | 17
b/res/res_pjsip_sdp_rtp.c | 124
b/res/res_pjsip_send_to_voicemail.c | 16
b/res/res_pjsip_session.c | 331
b/res/res_pjsip_session.exports.in | 1
b/res/res_pjsip_sips_contact.c | 107
b/res/res_pjsip_t38.c | 45
b/res/res_pjsip_transport_websocket.c | 146
b/res/res_pktccops.c | 6
b/res/res_rtp_asterisk.c | 416
b/res/res_security_log.c | 4
b/res/res_smdi.c | 4
b/res/res_sorcery_astdb.c | 1
b/res/res_sorcery_config.c | 71
b/res/res_sorcery_memory.c | 16
b/res/res_sorcery_memory_cache.c | 1059
b/res/res_sorcery_realtime.c | 16
b/res/res_stasis.c | 79
b/res/res_stasis_device_state.c | 54
b/res/res_stasis_playback.c | 8
b/res/res_stasis_recording.c | 10
b/res/res_stasis_snoop.c | 24
b/res/res_statsd.c | 88
b/res/res_statsd.exports.in | 4
b/res/res_timing_kqueue.c | 317
b/res/res_timing_pthread.c | 3
b/res/res_timing_timerfd.c | 5
b/res/res_xmpp.c | 38
b/res/snmp/agent.c | 10
b/res/stasis/app.c | 377
b/res/stasis/app.h | 15
b/res/stasis/control.c | 178
b/res/stasis/control.h | 8
b/res/stasis/messaging.c | 44
b/res/stasis/stasis_bridge.c | 60
b/res/stasis_recording/stored.c | 2
b/rest-api-templates/api.wiki.mustache | 20
b/rest-api-templates/ari.make.mustache | 1
b/rest-api-templates/ari_model_validators.c.mustache | 4
b/rest-api-templates/ari_resource.h.mustache | 19
b/rest-api-templates/asterisk_processor.py | 6
b/rest-api-templates/res_ari_resource.c.mustache | 71
b/rest-api-templates/swagger_model.py | 6
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 435
b/rest-api/api-docs/bridges.json | 6
b/rest-api/api-docs/channels.json | 101
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 6
b/rest-api/api-docs/events.json | 138
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/sounds/Makefile | 7
b/sounds/sounds.xml | 27
b/tests/test_acl.c | 45
b/tests/test_astobj2.c | 2
b/tests/test_cdr.c | 46
b/tests/test_cel.c | 50
b/tests/test_channel_feature_hooks.c | 4
b/tests/test_config.c | 174
b/tests/test_core_format.c | 105
b/tests/test_devicestate.c | 432
b/tests/test_dlinklists.c | 54
b/tests/test_expr.c | 2
b/tests/test_format_cap.c | 6
b/tests/test_func_file.c | 6
b/tests/test_gosub.c | 10
b/tests/test_message.c | 43
b/tests/test_pbx.c | 9
b/tests/test_poll.c | 2
b/tests/test_sched.c | 106
b/tests/test_sorcery.c | 132
b/tests/test_sorcery_memory_cache_thrash.c | 23
b/tests/test_sorcery_realtime.c | 12
b/tests/test_stasis.c | 4
b/tests/test_stasis_endpoints.c | 3
b/tests/test_stringfields.c | 2
b/tests/test_strings.c | 5
b/tests/test_threadpool.c | 60
b/tests/test_vector.c | 517
b/tests/test_voicemail_api.c | 12
b/third-party/Makefile | 21
b/third-party/Makefile.rules | 36
b/third-party/pjproject/.gitignore | 4
b/third-party/pjproject/Makefile | 145
b/third-party/pjproject/Makefile.rules | 7
b/third-party/pjproject/apply_patches | 39
b/third-party/pjproject/configure.m4 | 47
b/third-party/pjproject/patches/0001-2.4.5-fix-for-tls-async-ops.patch | 224
b/third-party/pjproject/patches/0001-Bump-tcp-tls-and-transaction-log-levels-from-1-to-3.patch | 70
b/third-party/pjproject/patches/0001-ioqueue-Enable-epoll-in-aconfigure.ac.patch | 80
b/third-party/pjproject/patches/0001-sip_transport-Search-for-transport-even-if-listener-.patch | 114
b/third-party/pjproject/patches/config_site.h | 34
b/third-party/pjproject/patches/user.mak | 2
b/third-party/versions.mak | 2
b/utils/.gitignore | 1
b/utils/Makefile | 3
b/utils/astman.c | 2
b/utils/clicompat.c | 7
b/utils/conf2ael.c | 13
b/utils/extconf.c | 336
contrib/ast-db-manage/config/versions/5a6ccc758633_add_pjsip_timeout_options.py | 24
contrib/asterisk-ng-doxygen | 1688
contrib/realtime/sqlserver/mssql_cdr.sql | 42
contrib/realtime/sqlserver/mssql_config.sql | 990
contrib/realtime/sqlserver/mssql_voicemail.sql | 48
res/res_jabber.exports.in | 16
res/res_pjsip_log_forwarder.c | 125
594 files changed, 64598 insertions(+), 20843 deletions(-)</pre><br></html>
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