Release Summary

asterisk-certified/16.3-cert1

Date: 2019-12-23

<asteriskteam@digium.com>


Table of Contents

  1. Summary
  2. Contributors
  3. Closed Issues
  4. Open Issues
  5. Other Changes
  6. Diffstat

Summary

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This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.

The data in this summary reflects changes that have been made since the previous release, asterisk-certified/13.21-cert6.


Contributors

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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.

CodersTestersReporters
693 Richard Mudgett
467 Corey Farrell
426 George Joseph
360 Joshua Colp
232 Matt Jordan
221 Alexander Traud
196 Sean Bright
183 Mark Michelson
127 Kevin Harwell
68 Alexei Gradinari (license 5691)
57 Rodrigo Ramírez Norambuena
43 Walter Doekes
38 Scott Griepentrog
36 Tzafrir Cohen
32 Jonathan Rose
31 Kinsey Moore
30 Torrey Searle
20 Ben Ford
19 David M. Lee
18 Diederik de Groot (License 6600)
16 Ivan Poddubny
13 Sungtae Kim
13 Diederik de Groot
13 Rusty Newton
12 Badalyan Vyacheslav
11 Timo Teräs
10 snuffy (license 5024)
10 Florian Floimair
10 Chris-Savinovich
9 Alexander Anikin
8 Matthew Fredrickson
8 Jaco Kroon
8 Daniel Journo
8 Jean Aunis
7 Aaron An
7 Martin Tomec
7 Benjamin Ford
7 Ashley Sanders
6 Joshua C. Colp
6 frahaase
6 Sebastian Gutierrez
6 sungtae kim
6 ibercom
6 Tyler Cambron
6 Joshua Elson
6 Michael Kuron
6 Guido Falsi
5 Kirill Katsnelson
5 Etienne Lessard
5 Gareth Palmer
5 Niklas Larsson
4 Emmanuel BUU
4 Jacek Konieczny
4 Michael Walton (license 6502)
4 Dade Brandon
4 Igor Goncharovskiy
4 Steve Davies
4 lvl
3 Daniel Tryba
3 Malcolm Davenport
3 Pascal Cadotte Michaud
3 Benjamin Keith Ford
3 Robert Mordec
3 Christof Lauber
3 Elazar Broad
3 Michael L. Young
3 Jason Parker (license 4993)
3 Ross Beer
3 Olle Johansson (License 5267)
3 Stefan Engström
3 Pirmin Walthert
3 Jeremy Laine
3 Richard Begg
3 Sergej Kasumovic
3 Jørgen H
3 Graham Barnett (License 6685)
3 abelbeck (License 5903)
3 Shaun Ruffell
3 Jeremy Lainé
3 Yousf Ateya
3 Scott Emidy
2 cirillor
2 Nuno Borges (License 6116)
2 Damian Ivereigh
2 David Hajek
2 Michael Cargile
2 Olle Johansson (License 5267)
2 Nathan Bruning
2 Nick French
2 sungtae kim
2 Benoît Dereck-Tricot
2 Asterisk Development Team
2 Francesco Castellano
2 Nir Simionovich
2 Gianluca Merlo
2 Richard Mudgett
2 Oron Peled
2 Y Ateya (License 6693)
2 Karsten Wemheuer
2 Philip Correia
2 Joerg Sonnenberger
2 Giuseppe Sucameli
2 yaron nahum (License 6676)
2 Nic Colledge
2 Sebastian Damm
2 Frederic LE FOLL
2 Andrew Nagy
2 Dennis Guse
2 cloos (License #5956)
2 Kristian Hogh (License 6639)
2 Richard Miller (license 5685)
2 Alec Davis
2 Vitezslav Novy
2 Kirsty Tyerman
2 John Bigelow
2 Igor Goncharovsky
2 Leif Madsen
1 Eugene
1 Maciej Szmigiero (license 6085)
1 Salah Ahmed
1 Sebastian Kemper
1 Zogot, cleaned up by me.
1 D Tucny
1 Justin T. Gibbs (License 6692)
1 Yasuhiko Kamata
1 Dmitry Bubnov (License 6651)
1 Filip Jenicek
1 nappsoft (license 6822)
1 Peter Katzmann (License 5968)
1 Vasil Kolev
1 Andreas Steinmetz (license 6523)
1 Andre Nazario
1 C.J. Collier
1 Bryan Boatright
1 Grachev Sergey
1 Krandon Bruse (license 6631)
1 Sam Wierema
1 demon-ru
1 Rodrigo Ramirez Norambuena (License 6577)
1 Jonh Wendell
1 Thierry Magnien
1 Robert Cripps
1 alex
1 Alexei Gradinari License #5691
1 Ludovic Gasc (GMLudo)
1 Alexandr Anikin
1 Ed Hynan (Licnese 6680)
1 Dömsödi Gergely
1 Michael Myles (License #6626)
1 Jonathan R. Rose
1 eyalhasson
1 Peter Katzmann
1 Jan Hoffmann (license 6986)
1 Eugene Voityuk
1 Nitesh Bansal (License #6418)
1 Roman S.
1 Yasin CANER
1 Jesper (License 5518)
1 Ben Smithurst (license 6529)
1 Ward van Wanrooij
1 Valentin Vidić (License 6697)
1 Dmitriy Bubnov (License 6651)
1 Joshua C. Colp
1 Olle E. Johansson
1 Matt Krokosz
1 Josh Roberson
1 Ryan Rittgarn
1 Kirsty Tyerman
1 Carlos Oliva
1 Eelco Brolman (License 6442)
1 Matt Hoskins (license 6688)
1 JoshE (license 6075)
1 Cao Minh Hiep
1 Chris Savinovich
1 Russell Bryant
1 Ben Klang (License 5876)
1 Makoto Dei (License 5027)
1 Di-Shi Sun (License 5076)
1 Evgeniy Tsybra
1 Eugene Voityuk
1 Xavier Hienne (License 6657)
1 Javier Acosta (License 6690)
1 Ian Gilmour (license 6889)
1 David Kerr
1 Xiemin Chen
1 Thomas Arimont (license 5525)
1 HZMI8gkCvPpom0tM (License 6658)
1 Alexander Traud
1 Dwayne Hubbard
1 LEI FU (License 6640)
1 chris de rock
1 Örn Arnarson
1 mdu113
1 Jan Juergens (License 6538)
1 Evandro Cesar Arruda
1 Gaurav Khurana
1 Corey Edwards
1 var
1 Kristian F. Høgh
1 Ben Merrills (License 6678)
1 server-pandora
1 Graham Mainwaring
1 Holger Hans Peter Freyther
1 Gerald Schnabel
1 Michael K (License 6621)
1 William McCall
1 Jasper Hafkenscheid
1 Badalian Vyacheslav (license 5249)
1 Damien Wedhorn
1 Peter Racz
1 Alexandre Fournier
1 Sergio Medina Toledo
1 Thomas Sevestre
1 Thomas Guebels
1 David J. Pryke
1 Mohit Dhiman
1 Chris Trobridge
1 Kristian Høgh (License #6639)
1 Birger Harzenetter (License 5870)
1 Eduardo S. Libardi
1 Roman Bedros (License 6842)
1 Stefan Engström (License 6691)
1 Lorenzo Miniero
1 Maciej Szmigiero
1 Mikheili Dautashvili
1 Norbert Varga
1 Brian P. Martin
1 Andrey Egorov
1 Michael K. (license 6621)
1 Javier Acosta
1 Paul Belanger
1 Simon Arlott (License 5756)
1 Leandro Dardini
1 gestoip2
1 Patric Marschall
1 Mark Duncan
1 Jan Friesse
1 Valentin Vidic
1 Moises Silva
1 Matthias Urlichs (license 5508)
1 Alessandro Crespi
1 David Duncan Ross Palmer (License 6660)
1 Debian Amtelco
1 Juergen Spies (License 6698)
1 Troy Bowman
1 Moritz Fain
1 Seán C McCord
1 Florian Sauerteig
1 Sebastien Duthil
76 George Joseph
6 Rusty Newton
5 AaronAn
4 Matt Jordan
3 Dmitry Melekhov
3 Etienne Lessard
3 Stefan Engström
3 Badalyan Vyacheslav
3 Emmanuel BUU
3 Alexander Traud
2 Aaron An
2 Michael L. Young
2 abelbeck
2 JoshE
2 Elazar Broad
2 snuffy
1 ibercom
1 Nick Adams
1 Alexandre Fournier
1 tootai
1 Sebastian Kemper
1 Andrew Nagy
1 opsmonitor
1 Arnd Schmitter
1 Zane Conkle
1 Dmitriy Serov
1 David J. Pryke
1 Samuel Galarneau
1 Walter Doekes
1 Yuriy Gorlichenko
1 starting asterisk -c until the colors stopped
1 Brad Latus
1 Graham Barnett
1 Brian Martin
1 Damian Ivereigh
1 XenCALL
1 Andrey Egorov
1 Beppo Maazucato
1 Ben Klang
1 Jacek Konieczny
1 dimitripietro
1 Ilya Shipitsin
1 Ivan Poddubny
1 Paolo Compagnini
1 Gregory Malsack
1 Damien Wedhorn
1 Ross Beer
1 Corey Edwards
1 David Hajek
1 Eugene Voityuk
1 Richard Mudgett
1 xrobau
1 Carl Fortin
1 Tony Lewis
1 David Duncan Ross Palmer
1 Deepak Singh Rawat
1 Juergen Spies
1 Shaun Ruffell
1 George Joseph
1 Cao Minh Hiep
1 Alexander Traud
1 Dan Cropp
1 Kilburn
1 Paul Belanger
1 David Herselman
1 Matt Hoskins
1 tests/test_utils.c.
1 Di-Shi Sun
1 Örn Arnarson
1 Ed Hynan
178 Alexander Traud
151 Matt Jordan
150 Corey Farrell
149 Joshua C. Colp
117 Richard Mudgett
92 George Joseph
73 Kevin Harwell
56 Mark Michelson
53 Alexei Gradinari
47 Mark Michelson
46 Richard Mudgett
41 Ross Beer
37 Tzafrir Cohen
31 Diederik de Groot
26 Torrey Searle
26 Scott Griepentrog
26 Rusty Newton
25 Etienne Lessard
25 Joshua Colp
25 Walter Doekes
21 Badalian Vyacheslav
21 sungtae kim
20 Rodrigo Ramirez Norambuena
19 Kevin Harwell
19 John Bigelow
19 Ross Beer
17 John Bigelow
16 George Joseph
14 Arnd Schmitter
14 Dmitriy Serov
14 Etienne Lessard
13 Jonathan Rose
13 Niklas Larsson
11 Rusty Newton
11 snuffy
11 Andrew Nagy
11 Stefan Engström
10 Sebastian Gutierrez
10 Sean Bright
10 Scott Griepentrog
9 Jean Aunis - Prescom
9 Tzafrir Cohen
9 Sandro Gauci
9 John Hardin
8 Andrew Nagy
8 Steve Pitts
8 abelbeck
8 Ashley Sanders
8 Ashley Sanders
7 Michael Maier
7 Jonathan Rose
7 David Brillert
7 Jeremy Lainé
7 Dennis Guse
7 Dan Jenkins
7 Joshua Elson
7 Richard Kenner
7 lvl
6 Sergej Kasumovic
6 Benjamin Keith Ford
6 Michael Keuter
6 Aaron An
6 JoshE
6 Badalian Vyacheslav
6 Niklas Larsson
6 Anthony Messina
6 Gareth Palmer
6 Morten Tryfoss
6 Nic Colledge
6 Jaco Kroon
6 yaron nahum
6 Guido Falsi
6 Alexander Traud
5 Frankie Chin
5 Dafi Ni
5 David M. Lee
5 yaron nahum
5 Marek Cervenka
5 Kirill Katsnelson
5 Dmitry Melekhov
5 Zane Conkle
5 Jonathan Harris
5 Richard Begg
5 Aaron An
5 Sandro Gauci
5 Michael Walton
5 Boris Fox
5 Dmitriy Serov
5 nappsoft
5 Dafi Ni
5 Carl Fortin
5 Florian Floimair
5 Zane Conkle
5 Gareth Palmer
5 Chet Stevens
4 Steve Davies
4 Richard Kenner
4 Vitezslav Novy
4 Marcello Ceschia
4 Javier Riveros
4 Ben Merrills
4 Jacek Konieczny
4 Emmanuel BUU
4 Ronald Raikes
4 Chet Stevens
4 Carl Fortin
4 Dade Brandon
4 John Nemeth
4 xrobau
4 Y Ateya
4 Kristian Høgh
4 Gianluca Merlo
4 dtryba
4 Walter Doekes
4 ibercom
4 Anthony Messina
4 Olle Johansson
4 Timo Teräs
4 Elazar Broad
4 Dmitry Melekhov
4 Michael Kuron
4 Leandro Dardini
4 Jørgen H
3 Richard Miller
3 Shaun Ruffell
3 Rodrigo Ramírez Norambuena
3 Igor Goncharovsky
3 Jesper
3 Ben Merrills
3 Ronald Raikes
3 Matthias Urlichs
3 tootai
3 Olle Johansson
3 Jeremy Kister
3 Jared Hull
3 Louis Jocelyn Paquet
3 Shaun Ruffell
3 Y Ateya
3 Daniel Journo
3 Marcello Ceschia
3 Graham Barnett
3 Javier Acosta
3 Marcelo Terres
3 Ian Gilmour
3 Ray Crumrine
3 Edwin Vandamme
3 Kirsty Tyerman
3 Olivier Krief
3 James Terhune
3 Private Name
3 Ray Crumrine
3 Matthias Urlichs
3 Tom Pawelek
3 Rodrigo Ramirez Norambuena
3 Frederic LE FOLL
3 Peter Katzmann
3 hristo
3 Jeremy Kister
3 Private Name
3 Sébastien Duthil
3 Emmanuel BUU
3 Jeremy Laine
3 Elazar Broad
3 Robert Mordec
3 Kinsey Moore
3 Kirsty Tyerman
3 Stefan Engström
2 warren smith
2 Xavier Hienne
2 Ben Smithurst
2 Samuel Galarneau
2 Steve Pitts
2 Gabriele Giacone <1o5g4r8o@gmail.com>
2 Denis Martinez
2 Daniel Heckl
2 Lorne Gaetz
2 Kristian Hogh
2 JoshE
2 Mitch Claborn
2 Ksenia
2 Josh Colp
2 HZMI8gkCvPpom0tM
2 Giuseppe Sucameli
2 Martin Cisárik
2 Cirillo Ferreira
2 Dan Jenkins
2 Daniel Heckl
2 Hans van Eijsden
2 nik600
2 Alexei Gradinari
2 John Nemeth
2 Harley Peters
2 Gergely Dömsödi
2 Sebastian Damm
2 Javier Acosta
2 Xavier Hienne
2 Sébastien Couture
2 Beppo Mazzucato
2 Alexandr Dranchuk
2 Sean Pimental
2 Ivan Poddubny
2 Michael K.
2 Thomas Thompson
2 Steven T. Wheeler
2 Max Norba
2 Brad Latus
2 Jesper
2 Krzysztof Trempala
2 Marcelo Terres
2 Frankie Chin
2 Evandro César Arruda
2 Nuno Borges
2 Ian Gilmour
2 David Hajek
2 twisted
2 HZMI8gkCvPpom0tM
2 mdu113
2 Ted G
2 cloos
2 Vadim
2 Kevin Scott Adams
2 Zach R
2 David Woolley
2 Carlos Chavez
2 Karsten Wemheuer
2 Makoto Dei
2 Stefan Repke
2 Marco Paland
2 Nuno Borges
2 Mitch Claborn
2 Abhay Gupta
2 David Kuehling
2 Thomas Frederiksen
2 Malcolm Davenport
2 seanchann.zhou
2 AaronAn
2 Michael
2 David Brillert
2 Jonathan R. Rose
2 Aleksei Kulakov
2 John Kiniston
2 Ove Aursand
2 David Woolley
2 Marco Giordani
2 Ben Smithurst
2 Diederik de Groot
2 Bryan Walters
2 Bradley Watkins
2 Ted G
2 Michael L. Young
2 Jens Bürger
2 Eyal Hasson
2 Steven Wheeler
2 Nathan Bruning
2 Graham Barnett
2 Jeffrey Walton
2 Filip Jenicek
2 Vitezslav Novy
2 Bojan Nemčić
2 John Zhong
2 Daniel Tryba
2 Damian Ivereigh
2 shaurya jain
2 Olivier Krief
2 Nir Simionovich (GreenfieldTech - Israel)
2 Damian Ivereigh
2 David Hajek
2 Taylor Hawkes
2 Florian Loyau
2 Kinsey Moore
2 Makoto Dei
2 Badalyan Vyacheslav
2 Patrick Laimbock
2 Alec Davis
2 klaus3000
2 Dmitry Wagin
2 Samuel Galarneau
2 PowerPBX
2 Philip Correia
2 Philip Correia
2 Christopher van de Sande
2 Bradley Watkins
2 Ilya Trikoz
2 Aaron Hamstra
2 Ben Klang
2 Nick French
2 Daniel Journo
2 Ludovic Gasc (Eyepea)
2 Sean Bright
2 Stuart Henderson
2 not here
1 Krandon Bruse
1 Maciej Szmigiero
1 Stephen More
1 Sebastian Kemper
1 Adam Secombe
1 Stefan Gofferje
1 Marcel Manz
1 Birger "WIMPy" Harzenetter
1 Stefan Gofferje
1 Harley Peters
1 Ivan Myalkin
1 Dmitriy Bubnov
1 Barry Chern
1 Thomas Sevestre
1 César Benjamín García Martínez
1 NITESH BANSAL
1 Niksa Baldun
1 Gareth Blades
1 dcarr
1 saghul
1 Ira Emus
1 Yasuhiko Kamata
1 PSDK
1 Timo Teräs
1 Matthias Binder
1 Seán C. McCord
1 Kevin McCoy
1 Peter Racz
1 Evers Lab
1 Per Jensen
1 Thiago Coutinho
1 Frank DiGennaro
1 David Kuehling
1 Michel R. Vaillancourt
1 Warren Selby
1 Yura Kocyuba
1 Michael K
1 Stephan Eisvogel
1 Leon Rowland
1 Bill Brigden
1 Dave Olszewski
1 Jason Richards
1 Frank DiGennaro
1 César Benjamín García Martínez
1 pasandev
1 John Covert
1 Dudás József
1 Salah Ahmed
1 Kevin McCoy
1 Paddy Grice
1 dimitripietro
1 Juris Breicis
1 Dmitriy Bubnov
1 'alex'
1 Michiel van Baak
1 Gregory Malsack
1 Daniel Flounders
1 Nick Ruggles
1 Ryan Rittgarn
1 Alex Villacís Lasso
1 Benoît Dereck-Tricot
1 David Duncan Ross Palmer
1 Andrew Zherdin
1 Ben Klang
1 Nir Simionovich
1 Sean Darcy
1 Luit van Drongelen
1 Abraham Liebsch
1 Roman Bedros
1 Valentin Safonov
1 Gaurav Khurana
1 Dmitry Burilov
1 Matt Krokosz
1 Dmitry Wagin
1 James Terhune
1 Bob Atkins
1 Lei Fu
1 Marco Giordani
1 XenCALL
1 Roy
1 Stephan Eisvogel
1 Yaniv Simhi
1 Dwayne Hubbard
1 Frederic Van Espen
1 Andrew Nowrot
1 Alexandre Fournier
1 Sotiris Ganouris
1 Denis Lebedev
1 Andrew Zherdin
1 Mark Petersen
1 Filip Frank
1 David Wilcox
1 Abhay Gupta
1 Florian Kaiser
1 David M. Lee
1 Ed Hynan
1 dea
1 Nic Colledge
1 Jesse Ross
1 Sebastian Damm
1 Anthony Critelli
1 Andreas Wetzel
1 Robert McGilvray
1 Lorne Gaetz
1 Yasin CANER
1 Marin Odrljin
1 Dmitriy
1 Roman Skvirsky
1 Brian Rel
1 Mohit Dhiman
1 Christoph Timm
1 Aleksei Kulakov
1 Leon Rowland
1 Nauman S
1 Nikolay shakin
1 Roman Bedros
1 Mateusz Kowalski
1 Andrey Egorov
1 Edwin Vandamme
1 Fran Vicente
1 Guido Falsi
1 Anatoli
1 Thomas Guebels
1 Simon Arlott
1 John Campbell
1 Marek Cervenka
1 Paul Belanger
1 Ross Beer, Jan Rozhon
1 Ilya Trikoz, Federico Santulli
1 Tim Morgan
1 Shane Blaser
1 Alex
1 Jacek Kowalski
1 Ryan Smith
1 Nicholas John Koch
1 Henning Holtschneider
1 Ustinov Artem
1 Jeppe Ryskov Larsen
1 Mark Thompson
1 Jason Richards
1 Javier Riveros
1 Martin Vit
1 Greg Siemon
1 Oleg Kozlov
1 Abraham Liebsch
1 LEI FU
1 ffs
1 Cameron
1 Dimos, Marco Giordani
1 Rustam Khankishyiev
1 Alejandro Mejia
1 Daniele Pallastrelli
1 effie mouzeli
1 Artur Pires
1 Michael L. Young
1 Josh Kitchens
1 Bryan Walters
1 Gil Richard
1 Brian Martin
1 Lorenzo Miniero
1 Avinash Mohod
1 WRP
1 Andreas Steinmetz
1 Martin Cisárik
1 Malcolm Davenport
1 Jeff Collell
1 Jens T.
1 Ilya Shipitsin
1 Alex A. Welzl
1 Andreas Steinmetz
1 Alexandr Dranchuk
1 Mr Dini
1 Ivan Ullmann
1 David Herselman
1 Bryant Zimmerman
1 Dimos
1 Vinod Dharashive
1 Ilya Shipitsin
1 Gil Richard
1 Jan Juergens
1 Mikhail
1 Thomas Frederiksen
1 Michelle Dupuis
1 Josh Kitchens
1 Michael Walton
1 gkloepfer
1 Sergio Medina Toledo
1 Leandro Dardini
1 Jonathan Cloots
1 warren smith
1 Krzysztof Trempala
1 Nick Repin
1 boatright
1 Benoît Dereck-Tricot
1 Matt DiMeo
1 Stefan27 (on IRC)
1 Barry Chern
1 Ward van Wanrooij
1 bautsche
1 Deepak Singh Rawat
1 Vasil Kolev
1 Smirnov Aleksey
1 Denis Alberto Martinez
1 Nasir Iqbal
1 József Dudás
1 Mark
1 Jim Van Meggelen
1 Arveno Santoro
1 David Cunningham
1 Tony Ching
1 Humberto Figuera
1 Robert Cripps
1 Николай Михо
1 Jared Hull
1 Peter Whisker
1 Cao Minh Hiep
1 feyfre
1 Eelco Brolman
1 Aaron Meriwether
1 Yaniv Simhi
1 Mateusz Kowalski
1 Carlos Oliva
1 Norbert Varga
1 Karsten Wemheuer
1 Stephane Chazelas
1 Igor Gamayunov
1 Beppo Mazzucato
1 Chris Howard
1 Andrey
1 Brian
1 Francesco Castellano
1 Vinod Dharashive
1 Eduardo Scudeller Libardi
1 Ben Langfeld
1 Alessandro Pimenta
1 Jacob Barber
1 Jatin Jain
1 Peter Sokolov
1 Said Masoud
1 Francois Blackburn
1 David Moore
1 Guenther Kelleter
1 Michael Newton
1 Dinis Brazão, Selene Feigl
1 Denis Lebedev
1 Paolo Compagnini
1 Sotiris Ganouris
1 Jens Bürger
1 Kilburn
1 Morton Tryfoss
1 M vd S
1 Rogger Padilla
1 Joel Vandal
1 Frederic LE FOLL
1 Bob Ham
1 David J. Pryke
1 C.J. Collier
1 basildane
1 Frederic Van Espen
1 Eliel Sardañons
1 Jonas Kellens
1 Sam Wierema
1 Nicolas Riendeau
1 Hiroaki Komatsu
1 Atis Lezdins
1 Melissa Shepherd
1 Roman Shubovich
1 Michael K.
1 Filip Jenicek
1 Richard Miller
1 Aleksandr Gordeev
1 Patric Marschall
1 Valentin Vidić
1 William McCall
1 Daniel Denson
1 Jay Jideliov
1 Rustam Khankishyiev
1 James Van Vleet
1 Charlie Smurthwaite
1 Guido Weckwerth
1 Martin Moučka
1 Michele Prà
1 James Van Vleet
1 Jared Biel
1 Eugene
1 Martin Tomec
1 Matt Hoskins
1 Nick Repin
1 David Moore
1 Terry Wilson
1 dkerr
1 Troy Bowman
1 Jeffrey Ollie
1 Shane Mitchell
1 Terry Wilson
1 Alessandro Polidori
1 cgi.net
1 Ross Beer.
1 Nicolas Riendeau
1 Bryant Zimmerman
1 Peter Katzmann
1 Pascal Cadotte Michaud
1 Stéphan Kochen
1 Michael Keuter
1 Hector Royo Concepcion
1 Humberto Figuera
1 Nick Ruggles
1 ibercom
1 Kilburn
1 Eduardo S. Libardi
1 Luit van Drongelen
1 Paul Sandys
1 Gerald Schnabel
1 Adam Secombe
1 Nasir Iqbal
1 Eric Dantie
1 Edvin Vidmar
1 Nicholas John Koch
1 Melissa Shepherd
1 Majdi Bsoul
1 Mark Thompson
1 Jacob Barber
1 Juan Sacco
1 Jim Van Meggelen
1 Jonathan R. Rose
1 dant
1 Allen Ford
1 Anatoli
1 Carlos Chavez
1 Samuel Owens
1 Benoit Duverger
1 Roman S.
1 Juergen Spies
1 Andreas Krüger
1 Ben Langfeld
1 Michele Prà
1 Ksenia
1 Gregory Malsack
1 Marian Koniuszko
1 Tony Mountifield
1 Huangyx
1 Tove Hjelm
1 Youngsung Kim at LINE Corporation
1 Ali Ghavidel
1 scgm11
1 Artem Volodin
1 Dominic
1 Eelco Brolman
1 Andre Nazario
1 Artem Volodin
1 Christoph Timm
1 Hunter Stevens, Said Masoud
1 Maxim Vasilev
1 Jared Biel
1 Vasilii Rogin
1 Hamid R. Hashmi
1 Adagio
1 Sebastian Gutierrez
1 Nick Adams
1 Halil İbrahim YILDIZ
1 Bojan Nemčić
1 Martin Moučka
1 Ray
1 CGI.NET
1 Marian Koniuszko
1 Aaron Meriwether
1 Sean McCord
1 jeffrey putnam
1 Francisco Seratti
1 Andrew Nowrot
1 Luke Hulsey
1 Jan Juergens
1 Michael Balen
1 John Fawcett
1 Patric Marschall
1 Yura Kocyuba
1 Andrey V. T.
1 Alexandr Gordeev
1 Thomas Airmont
1 Christopher van de Sande
1 Krandon Bruse
1 Kayode
1 Conrad de Wet
1 Roman Shubovich
1 Vitaly K
1 Torrey Searle, Nitesh Bansal
1 Matt Hoskins
1 Deepak Singh Rawat
1 Mak Dee
1 Matthew Fredrickson
1 Peter Racz
1 OpenBSD ports
1 Arnd Schmitter
1 Curt Sampson
1 Jens T.
1 David Justl
1 Tyler Cambron
1 Grigoriy Puzankin
1 seanchann.zhou
1 Dave Cabot
1 Birger Harzenetter
1 Frank Durden
1 Philippe Bolduc
1 John Kiniston
1 Maciej Szmigiero
1 Ivan Poddubny
1 Charlie Smurthwaite
1 Paul Sandys
1 Avinash Mohod
1 Curt Sampson
1 Nick Adams
1 Maxim Vasilev
1 Henning Holtschneider
1 Ivan Myalkin
1 Michael Myles
1 Shannon Price
1 Sean Darcy
1 Alec Davis
1 LEI FU
1 tm1000, Tony Lewis
1 clean targets.
1 Ustinov Artem
1 David Duncan Ross Palmer
1 Atis Lezdins
1 Eliel Sardañons
1 Hans van Eijsden
1 Mak Dee
1 Gergely Dömsödi
1 Marco Paland
1 Ed Hynan
1 Ryan Smith
1 Jan Hoffmann
1 John Harris
1 Tim Morgan
1 Max Man
1 David Cunningham
1 Darren Sessions
1 Simon Arlott
1 viniciusfontes
1 Ove Aursand
1 Stepan
1 Mark Petersen
1 Gareth Blades
1 Anthony Critelli
1 Allen Ford
1 Warren Selby
1 Panos Gkikakis
1 Grigoriy Puzankin
1 xiemchen
1 Holger Hans Peter Freyther
1 Shane Blaser
1 Sergey Grachev
1 Michael Myles
1 Stephen More
1 Jeffrey C. Ollie
1 Michael Cargile
1 Jan Hoffmann
1 Alex Odrov
1 John Covert
1 David Justl
1 Steve Murphy
1 rleasure
1 Will
1 Vitaly K
1 Matt Jordan III, Esq.
1 Cao Minh Hiep
1 Jacek
1 Wim De Vlaminck
1 Jeff Collell
1 John Zhong
1 Dave Olszewski
1 vadim
1 Francisco Seratti
1 Tim Ringenbach at Asteria Solutions Group
1 Andrey Biglari
1 Doug Lytle
1 Jeppe Ryskov Larsen
1 Halil İbrahim YILDIZ
1 chris de rock
1 Ruse
1 dhanapathy sathya
1 Damien Wedhorn, Matt Jordan
1 Andrej
1 wushumasters
1 StefanEng86, urbaniak, pay123
1 Lubos Dolezel
1 Greg Siemon
1 Hajek Michal
1 Dan Tucny
1 Jacques Peacock
1 Etienne Allovon
1 John Campbell
1 Barry Flanagan
1 Hector Royo Concepcion
1 Dwayne Hubbard
1 Siruja Maharjan
1 Conrad de Wet
1 Dwayne Hubbard
1 Örn Arnarson
1 Joerg Sonnenberger
1 Justin T. Gibbs
1 Edvin Vidmar
1 var
1 George Ladoff
1 Samuel For
1 Igor Gamayunov
1 Max Man
1 Darren Sessions
1 Philip Mott
1 David Herselman
1 David J. Pryke
1 Valentin Safonov
1 Stuart Henderson
1 Antoine Pitrou
1 Sebastian Kemper
1 Cyrille Demaret
1 Ivan Larionov
1 Graham Mainwaring
1 B. Davis
1 Rogger Padilla
1 Xiemin Chen
1 Alejandro Padilla
1 Jaco Kroon
1 Mark Scholten
1 Roman S.
1 Yaacov Akiba Slama
1 Kim youngsung
1 alex
1 Joerg Sonnenberger, D'Arcy Cain
1 Kristian Høgh
1 Brian J. Murrell
1 Tove Hjelm
1 Robert McGilvray
1 Örn Arnarson
1 John
1 Mikheili Dautashvili
1 Michael Newton
1 dhanapathy sathya
1 Jeremy Lainé
1 Ilya Trikoz
1 Chris Trobridge
1 David Wilcox
1 Brian
1 Andrew Green
1 John M.
1 Paddy Grice
1 Peter Whisker
1 jeffrey putnam
1 Patrick Laimbock
1 Janusz Karolak
1 Juergen Spies
1 Jonh Wendell
1 Jay Jideliov
1 Osaulenko Alexander
1 Kristijan Vrban
1 Cyril Ramière
1 Paul Belanger
1 Dave Cabot
1 cervajs, Inaki Baz Castillo

Closed Issues

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This is a list of all issues from the issue tracker that were closed by changes that went into this release.

Security

Category: Channels/chan_pjsip

ASTERISK-28260: Asterisk segfault when rtp negotiation is wrong or fails
Reported by: Sotiris GanourisASTERISK-27583: Segmentation fault occurs in asterisk with an invalid SDP fmtp attribute
Reported by: Sandro GauciASTERISK-27582: Segmentation fault occurs in Asterisk with an invalid SDP media format description
Reported by: Sandro GauciASTERISK-27640: SUBSCRIBE message with a large Accept value causes stack corruption
Reported by: Sandro Gauci

Category: Channels/chan_sip/General

ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter Addr of a peer
Reported by: Andrey V. T.

Category: Channels/chan_sip/Interoperability

ASTERISK-28465: Broken SDP can cause a segfault in a T.38 reINVITE
Reported by: Francesco Castellano

Category: Core/DNS

ASTERISK-28127: Buffer overflow for DNS SRV/NAPTR records
Reported by: Jan Hoffmann

Category: Core/HTTP

ASTERISK-27807: iostreams: Potential DoS when client connection closed prematurely
Reported by: Sean Bright

Category: Core/ManagerInterface

ASTERISK-28580: Bypass SYSTEM write permission in manager action allows system commands execution
Reported by: Eliel Sardañons

Category: Resources/res_http_websocket

ASTERISK-28013: res_http_websocket: Crash when reading HTTP Upgrade requests
Reported by: Sean BrightASTERISK-27658: WebSocket frames with 0 sized payload causes DoS
Reported by: Sean Bright

Category: Resources/res_pjsip

ASTERISK-27818: Username bruteforce is possible when using ACL with PJSIP
Reported by: John

Category: Resources/res_pjsip_messaging

ASTERISK-28447: res_pjsip_messaging: In-dialog MESSAGE with no body causes crash
Reported by: Gil Richard

Category: Resources/res_pjsip_t38

ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
Reported by: Alexei Gradinari

Category: pjproject/pjsip

ASTERISK-27618: Crash occurs when sending a repeated number of INVITE messages over TCP or TLS transport
Reported by: Sandro Gauci

New Feature

Category: Applications/NewFeature

ASTERISK-25551: [patch]Ability to add channel to an existing bridge by specifying an existing channel prefix
Reported by: Alec Davis

Category: Applications/app_chanspy

ASTERISK-25866: ChanSpy: allow usage of a long queue to store audio frames, to avoid audio loss
Reported by: Jean Aunis - Prescom

Category: Applications/app_confbridge

ASTERISK-25989: apps/confbridge: add regcontext feature
Reported by: Jaco Kroon

Category: Applications/app_controlplayback

ASTERISK-25654: Playback: Add the ability to play remote URIs
Reported by: Matt Jordan

Category: Applications/app_originate

ASTERISK-26587: app_originate: Add option to execute gosub prior to dial
Reported by: dkerr

Category: Applications/app_playback

ASTERISK-27286: Add the ability to read the media file type from HTTP header for playback
Reported by: Gaurav KhuranaASTERISK-25654: Playback: Add the ability to play remote URIs
Reported by: Matt Jordan

Category: Applications/app_queue

ASTERISK-26995: Add QUEUE_FLOAT_PENALTY to app_queue
Reported by: Steve DaviesASTERISK-19862: app_queue: Update Data of Queues (use queues as outbound calls container)
Reported by: Sebastian GutierrezASTERISK-16394: [patch] Last pause information to queue members
Reported by: Evandro César ArrudaASTERISK-25480: [patch]Add field PauseReason on QueueMemberStatus
Reported by: Rodrigo Ramirez NorambuenaASTERISK-23823: [patch] Option to keep queuerules in realtime
Reported by: Michael K.

Category: Applications/app_sms

ASTERISK-22591: [patch]Prevent Asterisk from writing received SMS content in log
Reported by: Jan Juergens

Category: Applications/app_voicemail

ASTERISK-17428: [patch] Allow "Comedian Mail" branding to be removed
Reported by: John CovertASTERISK-26087: Icelandic grammar support for voicemail and numbers
Reported by: Örn Arnarson

Category: CDR/NewFeature

ASTERISK-25479: Allow CDR's to be modified before being dispatched to engines
Reported by: Jonh Wendell

Category: CDR/cdr_adaptive_odbc

ASTERISK-25006: [patch] Add support set character for quoted identifiers
Reported by: Rodrigo Ramirez Norambuena

Category: CEL/cel_pgsql

ASTERISK-23186: [patch] Add usegmtime option to cel_pgsql
Reported by: Rodrigo Ramirez Norambuena

Category: Channels/General

ASTERISK-24363: [patch] Add ability for Channel Drivers to provide Presence State information
Reported by: Gareth Palmer

Category: Channels/chan_pjsip

ASTERISK-27478: PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI.
Reported by: Richard MudgettASTERISK-26277: Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
Reported by: Matt JordanASTERISK-25670: Add regcontext to PJSIP
Reported by: Daniel JournoASTERISK-25259: chan_pjsip: Add rtptimeout support
Reported by: Joshua C. ColpASTERISK-24341: PJSIP Ability to get info per contact
Reported by: xrobau

Category: Channels/chan_sip/General

ASTERISK-27163: chan_sip: Dialplan function SIP_HEADERS() to complement SIP_HEADER().
Reported by: Kirill Katsnelson

Category: Channels/chan_sip/Interoperability

ASTERISK-25803: [patch] chan_sip: Optionally supply fromuser/fromdomain in SIP dial string
Reported by: Walter Doekes

Category: Channels/chan_sip/NewFeature

ASTERISK-27162: [patch]chan_sip: Access incoming SIP REFER headers in the dialplan
Reported by: Kirill KatsnelsonASTERISK-17899: Handle crypto lifetime in SDES-SRTP negotiation
Reported by: Dwayne Hubbard

Category: Core/BuildSystem

ASTERISK-25591: [patch] Complete List of Header Files (#include): iwyu
Reported by: Alexander Traud

Category: Core/Channels

ASTERISK-27129: ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
Reported by: Corey Farrell

Category: Core/Configuration

ASTERISK-27117: core: Add support for timelen parsing to ast_parse_arg and ACO.
Reported by: Corey Farrell

Category: Core/General

ASTERISK-27413: Add cache_media_frames debugging option.
Reported by: Richard MudgettASTERISK-27063: Add support for systemd socket activation
Reported by: Corey FarrellASTERISK-26584: [patch] RTCP feedback for codec modules
Reported by: Lorenzo MinieroASTERISK-26630: Make logging PJPROJECT messages a bit easier
Reported by: Richard MudgettASTERISK-25419: Dialplan Application for Integration of StatsD
Reported by: Ashley SandersASTERISK-24834: DNS Overhaul: Implement the proposed core API - sync/async functions, resolver registration
Reported by: Matt JordanASTERISK-24836: DNS Overhaul: Write a Resolver Implementation
Reported by: Matt Jordan

Category: Core/HTTP

ASTERISK-27063: Add support for systemd socket activation
Reported by: Corey Farrell

Category: Core/Logging

ASTERISK-25425: logger: Add JSON structured logging
Reported by: Matt Jordan

Category: Core/ManagerInterface

ASTERISK-27215: [patch]AMI : Add CancelAtxfer Action
Reported by: Thomas SevestreASTERISK-27063: Add support for systemd socket activation
Reported by: Corey FarrellASTERISK-26058: [Patch] Add uptime and last reloaded to FullyBooted AMI event
Reported by: Niklas LarssonASTERISK-24554: AMI/ARI: Generate events on connected line changes
Reported by: Matt Jordan

Category: Core/ManagerInterface/NewFeature

ASTERISK-25904: PJSIP: add contact.updated event
Reported by: Alexei GradinariASTERISK-25903: PJSIP AMI Event ContactStatus: add Useragent and RegExpire
Reported by: Alexei Gradinari

Category: Core/Netsock

ASTERISK-27063: Add support for systemd socket activation
Reported by: Corey Farrell

Category: Core/NewFeature

ASTERISK-27413: Add cache_media_frames debugging option.
Reported by: Richard MudgettASTERISK-24363: [patch] Add ability for Channel Drivers to provide Presence State information
Reported by: Gareth Palmer

Category: Features

ASTERISK-27215: [patch]AMI : Add CancelAtxfer Action
Reported by: Thomas Sevestre

Category: Formats/NewFeature

ASTERISK-18995: Support for OGG/Speex file format
Reported by: Timo Teräs

Category: Functions/func_channel

ASTERISK-26878: func_channel: Add ability to get the callid so dialplan has access to it.
Reported by: Richard Mudgett

Category: Functions/func_curl

ASTERISK-25652: func_curl: Add the ability to CURL files down to a specified location
Reported by: Matt Jordan

Category: General

ASTERISK-26595: ARI: Add the ability to control the source of video in a multi-party mixing bridge
Reported by: Matt JordanASTERISK-26470: ARI: Add an 'asterisk_id' field to outgoing events
Reported by: Matt JordanASTERISK-26087: Icelandic grammar support for voicemail and numbers
Reported by: Örn ArnarsonASTERISK-26068: Multicast RTP Options
Reported by: Mark MichelsonASTERISK-25972: res_pjsip_exten_state: Use body generator to publish extension state
Reported by: Richard MudgettASTERISK-25889: ARI: Add separate "create" and "dial" operations for channels
Reported by: Mark MichelsonASTERISK-25660: Add sipp-sendfax.xml and spandspflow2pcap.py to contrib/scripts.
Reported by: Walter DoekesASTERISK-25549: Confbridge: Add participant timeout option
Reported by: Mark MichelsonASTERISK-24931: dns: Add support for SRV records.
Reported by: Joshua C. ColpASTERISK-23871: RLS Tests: Implement RLS off-nominal tests
Reported by: Mark Michelson

Category: PBX/NewFeature

ASTERISK-27162: [patch]chan_sip: Access incoming SIP REFER headers in the dialplan
Reported by: Kirill Katsnelson

Category: Resources/res_ari

ASTERISK-28267: res_stasis: Add ability to switch applications
Reported by: Benjamin Keith FordASTERISK-27322: [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge
Reported by: Darren SessionsASTERISK-26492: ARI: Add ability to specify channel variables on websocket events
Reported by: Mark MichelsonASTERISK-25925: Allow Early Bridges on ARI Dials
Reported by: Mark MichelsonASTERISK-26022: ARI: Add media playlists
Reported by: Matt JordanASTERISK-25252: ARI: Add the ability to manipulate log channels
Reported by: Matt JordanASTERISK-25238: ARI: Support push configuration
Reported by: Matt JordanASTERISK-25173: ARI: Add the ability to load/reload/unload an Asterisk module
Reported by: Matt JordanASTERISK-24554: AMI/ARI: Generate events on connected line changes
Reported by: Matt Jordan

Category: Resources/res_ari_bridges

ASTERISK-26022: ARI: Add media playlists
Reported by: Matt Jordan

Category: Resources/res_ari_channels

ASTERISK-26022: ARI: Add media playlists
Reported by: Matt JordanASTERISK-24922: ARI: Add the ability to intercept hold and raise an event
Reported by: Matt JordanASTERISK-24703: ARI: Add the ability to "transfer" (redirect) a channel
Reported by: Matt Jordan

Category: Resources/res_ari_recordings

ASTERISK-26042: ARI: Allow downloading of the media associated with a stored recording
Reported by: Matt Jordan

Category: Resources/res_musiconhold

ASTERISK-24276: [Patch] Option to make app MOH override channel musicclass
Reported by: Kristian Høgh

Category: Resources/res_pjsip

ASTERISK-27704: Add cache_pools debug option to pjproject.conf
Reported by: Richard MudgettASTERISK-27581: Add new AMI Action for PJSIPShowContacts
Reported by: sungtae kimASTERISK-27547: res_pjsip: Add new AMI Action for PJSIPShowAuths
Reported by: sungtae kimASTERISK-27478: PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI.
Reported by: Richard MudgettASTERISK-27206: res_pjsip: No mechanism exists to limit endpoint identification to IP only
Reported by: Ben MerrillsASTERISK-26863: res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
Reported by: Matt JordanASTERISK-25904: PJSIP: add contact.updated event
Reported by: Alexei GradinariASTERISK-25900: PJSIP Endpoint IP Access Controls
Reported by: Alexei GradinariASTERISK-25903: PJSIP AMI Event ContactStatus: add Useragent and RegExpire
Reported by: Alexei GradinariASTERISK-24919: res_pjsip_config_wizard: Ability to write contents to file
Reported by: Ray CrumrineASTERISK-25377: res_pjsip: Change default "From user" from UUID to something more palatable
Reported by: Mark Michelson

Category: Resources/res_pjsip/Bundling

ASTERISK-26630: Make logging PJPROJECT messages a bit easier
Reported by: Richard Mudgett

Category: Resources/res_pjsip_outbound_publish

ASTERISK-25901: Add transport for outbound PUBLISH
Reported by: Alexei Gradinari

Category: Resources/res_pjsip_sdp_rtp

ASTERISK-25259: chan_pjsip: Add rtptimeout support
Reported by: Joshua C. Colp

Category: Resources/res_pjsip_session

ASTERISK-28087: add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip
Reported by: Torrey SearleASTERISK-27478: PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE Request-URI.
Reported by: Richard Mudgett

Category: Resources/res_rtp_asterisk

ASTERISK-26584: [patch] RTCP feedback for codec modules
Reported by: Lorenzo Miniero

Category: Resources/res_stasis

ASTERISK-27322: [New Feature] Add mute and DTMF passthrough to ARI add channel to bridge
Reported by: Darren Sessions

Category: Resources/res_statsd

ASTERISK-25419: Dialplan Application for Integration of StatsD
Reported by: Ashley Sanders

Bug

Category: . I did not set the category correctly.

ASTERISK-28221: Bug in ast_coredumper
Reported by: Andrew NagyASTERISK-27878: [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-26391: Consoles do not display verbose logger messages even when requested.
Reported by: Marcelo TerresASTERISK-24147: ARI: channel hangup crashes asterisk process
Reported by: Edvin Vidmar

Category: .Release/Targets

ASTERISK-27800: One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP
Reported by: Artur Pires

Category: Addons/General

ASTERISK-25640: pbx: Deadlock on features reload and state change hint.
Reported by: Krzysztof Trempala

Category: Addons/cdr_mysql

ASTERISK-27572: cdr_mysql creates empty records if reconnects when mysql was not up on module load
Reported by: Tzafrir CohenASTERISK-27782: cdr_mysql: Missing MYSQL_PORT definition
Reported by: Evandro César ArrudaASTERISK-27366: Asterisk Turkish Language Set Problem
Reported by: Halil İbrahim YILDIZASTERISK-27270: cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
Reported by: Tzafrir CohenASTERISK-26359: [patch] cdr_mysql: fails to use UTC if so instructed
Reported by: Tzafrir Cohen

Category: Addons/chan_mobile

ASTERISK-27726: chan_mobile: presents incorrect inbound Caller-ID names
Reported by: BrianASTERISK-24468: Incoming UCS2 encoded SMS truncated if SMS length exceeds 50 (roughly) national symbols
Reported by: Dmitriy Bubnov

Category: Addons/chan_ooh323

ASTERISK-27938: [patch] Compile fails with `IPTOS_MINCOST' undeclared.
Reported by: Alexander TraudASTERISK-27901: [patch] ooh323c: GCC 8: output truncated before terminating nul.
Reported by: Alexander TraudASTERISK-27812: When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk.
Reported by: DimosASTERISK-26893: No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module
Reported by: Marco GiordaniASTERISK-27577: [patch] chan_ooh323: Avoid typecasting an int to unsigned short.
Reported by: Alexander TraudASTERISK-27557: [patch] clang 5.0: implicit conversion to char changes value to negative.
Reported by: Alexander TraudASTERISK-27552: [patch] chan_ooh323: Limit outgoinglimit to positive values as intended.
Reported by: Alexander TraudASTERISK-27551: [patch] ooh323cDriver: Fix typo in header guard.
Reported by: Alexander TraudASTERISK-27353: H323 audio starts with a delay of 2 seconds.
Reported by: Marco GiordaniASTERISK-24400: ooh323 sends wrong hangup code
Reported by: Dmitry MelekhovASTERISK-25227: No audio at in-band announcements in ooh323 channel
Reported by: Alexandr DranchukASTERISK-25299: RTP port leaks with incoming OOH323 calls
Reported by: Alexandr DranchukASTERISK-24393: rtptimeout=0 doesn't disable rtptimeout
Reported by: Dmitry Melekhov

Category: Addons/format_mp3

ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
Reported by: Tzafrir Cohen

Category: Addons/res_config_mysql

ASTERISK-27607: [patch] res_config_mysql: Avoid the header mysql_version.h.
Reported by: Alexander TraudASTERISK-18271: Pattern matching with res_config_mysql extensions does not behave as expected
Reported by: Charlie SmurthwaiteASTERISK-26362: res_config_mysql: Broken after 13.10
Reported by: Carlos ChavezASTERISK-18252: queue_log mysql time column data format
Reported by: Gareth BladesASTERISK-25041: [patch]Broken column type checking in res_config_mysql addon
Reported by: Alexandre Fournier

Category: Applications/General

ASTERISK-26997: Create an StreamEcho dialplan application
Reported by: Kevin Harwell

Category: Applications/app_adsiprog

ASTERISK-27557: [patch] clang 5.0: implicit conversion to char changes value to negative.
Reported by: Alexander Traud

Category: Applications/app_agent_pool

ASTERISK-24737: When agent not logged in, agent status shows unavailable, queue status shows agent invalid
Reported by: Richard MudgettASTERISK-24257: agent must dial acceptdtmf twice to bridge to queue caller
Reported by: Steve Pitts

Category: Applications/app_amd

ASTERISK-27610: app_amd.so returning TOOLONG before reaching the timeout
Reported by: Michael CargileASTERISK-25639: app_amd: system maxwords discrepency
Reported by: Dade BrandonASTERISK-19470: Documentation on app_amd is incorrect
Reported by: Frank DiGennaro

Category: Applications/app_chanspy

ASTERISK-25321: [patch]DeadLock ChanSpy with call over Local channel
Reported by: Filip FrankASTERISK-25247: choppy audio when spying on a g722 channel, chan_sip or chan_pjsip
Reported by: hristoASTERISK-24828: Fix Frame Leaks
Reported by: Kevin Harwell

Category: Applications/app_confbridge

ASTERISK-28201: [patch] confbridge: no announce to the marked users when they join an empty conference
Reported by: Alexei GradinariASTERISK-28107: app_confbridge: Participant info labels aren't being added to the SDPs
Reported by: George JosephASTERISK-27870: app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts
Reported by: Robert MordecASTERISK-27804: bridge_softmix / app_confbridge: Add support for combining REMB reports
Reported by: Joshua C. ColpASTERISK-27418: app_confbridge: "core show profile bridge" does not output "sfu" when video_mode is sfu
Reported by: Carlos ChavezASTERISK-27786: app_confbridge: Add ability to enable and configure REMB support
Reported by: Joshua C. ColpASTERISK-27755: ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status
Reported by: Kevin HarwellASTERISK-24756: ConfBridge sound_muted does not work from CLI or AMI
Reported by: Thomas FrederiksenASTERISK-27378: Modules: Fix issues with CLI completion.
Reported by: Corey FarrellASTERISK-26994: Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
Reported by: James TerhuneASTERISK-27123: confbridge: Name recordings are left on filesystem
Reported by: Sergej KasumovicASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user name recording while leaving
Reported by: Robert MordecASTERISK-25506: [patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
Reported by: Frederic LE FOLLASTERISK-20987: non-admin users, who join muted conference are not being muted
Reported by: hristoASTERISK-25253: confbridge volume options and other volume controls such as func_volume don't work
Reported by: Dmitriy SerovASTERISK-24749: ConfBridge: Wrong language on playing conf-hasjoin and conf-hasleft when played to bridge
Reported by: Philippe BolducASTERISK-24841: ConfBridge: Strange sampling rates chosen when channels have multiple native formats
Reported by: Matt JordanASTERISK-24864: app_confbridge: file playback blocks dtmf
Reported by: Kevin HarwellASTERISK-24719: ConfBridge recording channels get stuck when recording started/stopped more than once
Reported by: Richard MudgettASTERISK-24723: confbridge: CLI command 'confbridge list XXXX' no longer displays user menus
Reported by: Matt JordanASTERISK-24490: Security Vulnerability: CONFBRIDGE function's record_command option allows arbitrary parameters to be passed to MixMonitor, allowing remote execution of commands
Reported by: Matt JordanASTERISK-24522: ConfBridge: delay occurs between kicking all endmarked users when last marked user leaves
Reported by: Matt JordanASTERISK-22409: Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade
Reported by: Corey FarrellASTERISK-24208: Channels with CDR Information Remain Active Even After ConfBrige Is Ended
Reported by: Frankie Chin

Category: Applications/app_controlplayback

ASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Reported by: Matt Jordan

Category: Applications/app_dial

ASTERISK-27980: Caller ID cannot be changed on Attended Transfer before dialing out
Reported by: Alexei GradinariASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Reported by: Rusty NewtonASTERISK-26549: app_dial: When PickupChan() is used some channels may have incorrect device state
Reported by: Joshua C. ColpASTERISK-26446: app_dial: There's no way to override the hangupcause on unanswered channels
Reported by: George JosephASTERISK-25691: Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
Reported by: Etienne LessardASTERISK-26282: AEL: macro-call in Dial application, macro "lacks 's' extension"
Reported by: chris de rockASTERISK-24958: Forwarding loop detection inhibits certain desirable scenarios
Reported by: Mark MichelsonASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard MudgettASTERISK-25212: [patch]Segfault when using DEBUG_FD_LEAKS
Reported by: Walter DoekesASTERISK-24682: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value
Reported by: Matt JordanASTERISK-24138: dial: Call forwarding information presented through AMI/ARI is wrong
Reported by: Matt JordanASTERISK-24225: Dial option z is broken
Reported by: dimitripietro

Category: Applications/app_directory

ASTERISK-27241: libc segfault upon entry into app_directory
Reported by: David MooreASTERISK-27093: ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
Reported by: James TerhuneASTERISK-25087: Asterisk segfault when using Directory application with alias option and specific mailbox configuration
Reported by: Chet Stevens

Category: Applications/app_echo

ASTERISK-25867: [patch] Video delay on app_echo
Reported by: Jacek Konieczny

Category: Applications/app_fax

ASTERISK-27671: Deprecate legacy modules
Reported by: Corey Farrell

Category: Applications/app_followme

ASTERISK-27980: Caller ID cannot be changed on Attended Transfer before dialing out
Reported by: Alexei GradinariASTERISK-26288: followme: fails to reset config items to default values on reload
Reported by: Tzafrir CohenASTERISK-26008: app_followme does not delete recorded name prompt
Reported by: Tzafrir Cohen

Category: Applications/app_macro

ASTERISK-26570: Macro allows an infinite loop of dialplan inclusion resulting in a crash
Reported by: Tzafrir CohenASTERISK-27350: app_macro deprecation
Reported by: Corey FarrellASTERISK-26282: AEL: macro-call in Dial application, macro "lacks 's' extension"
Reported by: chris de rock

Category: Applications/app_meetme

ASTERISK-28328: MeetMe global non-admin mute is muting admins that subsequently join
Reported by: Philip MottASTERISK-27378: Modules: Fix issues with CLI completion.
Reported by: Corey FarrellASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua C. ColpASTERISK-25569: app_meetme: Audio quality issues
Reported by: Corey FarrellASTERISK-24572: [patch]App_meetme is loaded without its defaults when the configuration file is missing
Reported by: Nuno BorgesASTERISK-24234: app_meetme: Crash on conference shutdown due to NULL channel passed to meetme_stasis_generate_msg()
Reported by: Shaun Ruffell

Category: Applications/app_minivm

ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-20858: app_minivm fails to clean up mkstemp files
Reported by: Walter Doekes

Category: Applications/app_mixmonitor

ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-26169: format_ogg_vorbis: Memory leak using OGG in MixMonitor
Reported by: Ivan MyalkinASTERISK-26875: app_mixmonitor: Recording out of sync when 183 but no RTP
Reported by: Aaron AnASTERISK-26867: autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
Reported by: Krzysztof TrempalaASTERISK-21094: MixMonitorMute mutes through stream if already slinear (e.g. Originate)
Reported by: David WoolleyASTERISK-25322: Crash occurs when using MixMonitor with t() or r() options.
Reported by: Richard MudgettASTERISK-24195: bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge
Reported by: Jonathan RoseASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Reported by: Matt Jordan

Category: Applications/app_mp3

ASTERISK-26085: app_mp3: results in timeout for streams
Reported by: Jens Bürger

Category: Applications/app_originate

ASTERISK-25266: Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
Reported by: Allen Ford

Category: Applications/app_osplookup

ASTERISK-27578: [patch] app_osplookup.c: Avoid a format truncation.
Reported by: Alexander Traud

Category: Applications/app_page

ASTERISK-25384: Regular Asterisk crashes when using Page application. "user_data is NULL"
Reported by: Chet Stevens

Category: Applications/app_playback

ASTERISK-27124: app_playback.c:say_date_generic use timezonename parameter
Reported by: Holger Hans Peter FreytherASTERISK-26774: core: Playback URL fails after some time
Reported by: Igor Gamayunov

Category: Applications/app_queue

ASTERISK-27541: app_queue: Queue paused reason was (big number) secs ago when reason is set
Reported by: César Benjamín García MartínezASTERISK-20986: QUEUE_MEMBER 's description is inaccurate
Reported by: Olivier KriefASTERISK-27964: app_queue: ring_entry accesses nativeformats without channel lock or reference
Reported by: Francisco SerattiASTERISK-28168: app_queue: Adding a blank entry into sql queue_members crashes asterisk.
Reported by: MichaelASTERISK-28218: app_queue: Asterisk crashes when using Queue with a pre-dial handler (option b)
Reported by: MarkASTERISK-28125: app_queue: Revert broken queue channel reference patch
Reported by: lvlASTERISK-27980: Caller ID cannot be changed on Attended Transfer before dialing out
Reported by: Alexei GradinariASTERISK-27920: app_queue: Queue member considered inuse after immediately hanging up during dialing.
Reported by: Cao Minh HiepASTERISK-28032: Realtime queuemembers are not updated during retry phase
Reported by: lvlASTERISK-27973: app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
Reported by: Valentin SafonovASTERISK-18411: Queue members with hints for state_interface get stuck in "In Use" state.
Reported by: Steven WheelerASTERISK-27301: [patch] app_queue: Music On Hold for real-time queues is not reset to default
Reported by: Nathan BruningASTERISK-27216: app_queue: does its check-makeannouncement-logic twice each head-caller-loop
Reported by: Stefan EngströmASTERISK-27232: When in queue on g722 with interruptions, music on hold can get stuck and no longer play
Reported by: Jens T.ASTERISK-19103: When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
Reported by: Jim Van MeggelenASTERISK-27204: [patch] app_queue: Wrong queue stat calculation
Reported by: sungtae kimASTERISK-27073: manager: AMI "queues" action outputs freeform text that doesn't follow the AMI spec
Reported by: BrianASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove AursandASTERISK-27065: call hangup after leaving app_queue
Reported by: Marek CervenkaASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumastersASTERISK-26400: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
Reported by: Etienne LessardASTERISK-26715: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
Reported by: David BrillertASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
Reported by: Lorne GaetzASTERISK-26920: app_queue: PAUSEALL/UNPAUSEALL does not log reason
Reported by: Troy BowmanASTERISK-26862: app_queue: Queue stops calling members with local interface after forwarding in previous call
Reported by: Robert MordecASTERISK-23457: SQlite3: Realtime queue loading fails after PRAGMA query result
Reported by: Scott GriepentrogASTERISK-26775: app_queue: reset abandoned in service level
Reported by: Sebastian GutierrezASTERISK-26755: app_queue: Random queues disappear on "core reload queue all"
Reported by: Kirill KatsnelsonASTERISK-26665: app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
Reported by: Marek CervenkaASTERISK-26621: app_queue: Queue application does not ring members with Local interface
Reported by: Jonas KellensASTERISK-26462: [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
Reported by: Leandro DardiniASTERISK-26330: app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
Reported by: Etienne LessardASTERISK-26360: app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
Reported by: Richard MudgettASTERISK-26299: app_queue: Queue application sometimes stops calling members with Local interface
Reported by: Etienne LessardASTERISK-25797: app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
Reported by: Etienne LessardASTERISK-26133: app_queue: Queue members receive multiple calls
Reported by: Richard MillerASTERISK-16115: [patch] problem with ringinuse=no, queue members receive sometimes two calls
Reported by: nik600ASTERISK-25954: Manager QueueSummary and QueueStatus Actions are case sensitive to QueueName
Reported by: Javier AcostaASTERISK-25888: Frequent segfaults in function can_ring_entry() of app_queue.c
Reported by: Sébastien CoutureASTERISK-25800: [patch] Calculate talktime when is first call answered
Reported by: Rodrigo Ramirez NorambuenaASTERISK-25732: [patch] persist queue member pause reason through restart
Reported by: Rodrigo Ramirez NorambuenaASTERISK-19820: wrapuptime is intermittently disregarded for queue calls
Reported by: WRPASTERISK-25442: using realtime (mysql) queue members are never updated in wait_our_turn function (app_queue.c)
Reported by: Carlos OlivaASTERISK-25561: app_queue.c line 6503 (try_calling): mutex 'qe->chan' freed more times than we've locked!
Reported by: Alec DavisASTERISK-25423: Caller gets no Connected line update during call pickup.
Reported by: Richard MudgettASTERISK-25399: app_queue: AgentComplete event has wrong reason
Reported by: Kevin HarwellASTERISK-25185: Segfault in app_queue on transfer scenarios
Reported by: Etienne LessardASTERISK-25215: Differences in queue.log between Set QUEUE_MEMBER and using PauseQueueMember
Reported by: Lorne GaetzASTERISK-25038: Queue log "EXITWITHTIMEOUT" does not always contain waiting time
Reported by: Etienne LessardASTERISK-23319: Segmentation fault in queue_exec at app_queue.c
Reported by: VadimASTERISK-24267: Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel
Reported by: Mitch ClabornASTERISK-24466: app_queue: fix a couple leaks to struct call_queue
Reported by: Corey FarrellASTERISK-24454: app_queue: ao2_iterator not destroyed, causing leak
Reported by: Corey Farrell

Category: Applications/app_record

ASTERISK-27423: app_record: We set the RECORD_STATUS channel variable before closing the file
Reported by: George JosephASTERISK-16777: several filename bugs in Record() application
Reported by: klaus3000ASTERISK-18286: [patch] 'Silence' is truncated in Record()
Reported by: varASTERISK-25410: app_record: RECORDED_FILE variable not being populated
Reported by: Kevin Harwell

Category: Applications/app_saynumber

ASTERISK-26598: Saynumber is trying to get "and" from "digits/" subfolder
Reported by: Jonathan Harris

Category: Applications/app_sayunixtime

ASTERISK-25810: say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
Reported by: Nicolas Riendeau

Category: Applications/app_sms

ASTERISK-27557: [patch] clang 5.0: implicit conversion to char changes value to negative.
Reported by: Alexander Traud

Category: Applications/app_stasis

ASTERISK-26716: ari: Channels with pre-dial handlers cannot be hung up via ARI
Reported by: Tom Pawelek

Category: Applications/app_system

ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell

Category: Applications/app_transfer

ASTERISK-25649: Transfer application does not work with Local channels - documentation misleading
Reported by: Ivan UllmannASTERISK-24015: app_transfer fails with PJSIP channels
Reported by: Private Name

Category: Applications/app_voicemail

ASTERISK-28306: res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent
Reported by: Jared HullASTERISK-28166: app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC
Reported by: MichaelASTERISK-28225: app_voicemail: Channel variable VM_MESSAGEFILE not updated correctly if message marked "urgent"
Reported by: boatrightASTERISK-28222: Regression: MWI polling no longer works
Reported by: abelbeckASTERISK-28215: app_voicemail: Leaving voicemail sometimes doesn't trigger NOTIFYs
Reported by: George JosephASTERISK-28151: app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default
Reported by: Ronald RaikesASTERISK-27853: Incorrect error reported when leaving/retrieving a ODBC voicemail
Reported by: Nic ColledgeASTERISK-27703: AMI Action VoicemailUsersList returns 0 MessageCount
Reported by: Sébastien DuthilASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-21241: When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
Reported by: Eelco BrolmanASTERISK-27171: Asterisk 15.0.0-Beta1 does not compile
Reported by: Ira EmusASTERISK-24562: app_voicemail: Cannot set fromstring on a per-mailbox basis
Reported by: Mark ScholtenASTERISK-25893: Function vmauthenticate accesses uninitialized memory
Reported by: Filip JenicekASTERISK-26723: VoiceMailPlayMsg not playing messages via realtime
Reported by: Ryan RittgarnASTERISK-26503: app_voicemail: Asterisk crashes when MailboxExists is used
Reported by: Doug LytleASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code.
Reported by: Corey FarrellASTERISK-26045: [patch]app_voicemail: fix bugs, imap mm_status log change to debug
Reported by: Alexei GradinariASTERISK-24463: Voicemail email address corrupt or not sent when message is in the process of being recorded during reload
Reported by: John CampbellASTERISK-25917: [patch]app_voicemail: passwordlocation=spooldir only works if you manually add secret.conf yourself
Reported by: Jonathan R. RoseASTERISK-25874: app_voicemail: Stack buffer overflow in test_voicemail_notify_endl
Reported by: Badalian VyacheslavASTERISK-25082: Asterisk deletes message after doing a playback of an INBOX message using ast_vm_play when the Old folder is full for that mailbox.
Reported by: Jonathan RoseASTERISK-24626: Voicemail passwords not being stored in ARA
Reported by: Paddy GriceASTERISK-24709: [patch] msg_create_from_file used by MixMonitor m() option does not queue an MWI event
Reported by: Gareth PalmerASTERISK-24250: [patch] Voicemail with multi-recipients To: header fix
Reported by: abelbeckASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks
Reported by: Corey FarrellASTERISK-24190: IMAP voicemail causes segfault
Reported by: Nick Adams

Category: Applications/app_voicemail/IMAP

ASTERISK-27639: [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD.
Reported by: Alexander TraudASTERISK-27734: [patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux.
Reported by: Alexander TraudASTERISK-27681: [patch] BuildSystem: Enable IMAP storage on OpenBSD.
Reported by: Alexander TraudASTERISK-27635: [patch] app_voicemail: Avoid always true warnings with clang.
Reported by: Alexander TraudASTERISK-27181: GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting'
Reported by: Anthony MessinaASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets.
Reported by: Louis Jocelyn PaquetASTERISK-26045: [patch]app_voicemail: fix bugs, imap mm_status log change to debug
Reported by: Alexei GradinariASTERISK-24927: app_voicemail (IMAP support) function save_to_folder: creates wrong folder
Reported by: Alexei GradinariASTERISK-25899: IMAP access FATAL error: Out of memory
Reported by: Alexei GradinariASTERISK-24786: [patch] - Asterisk terminates when playing a voicemail stored in LDAP
Reported by: Graham BarnettASTERISK-24787: [patch] - Microsoft exchange incompatibility for playing back messages stored in IMAP - play_message: No origtime
Reported by: Graham BarnettASTERISK-24288: [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
Reported by: LEI FUASTERISK-24516: [patch]Asterisk segfaults when playing back voicemail under high concurrency with an IMAP backend
Reported by: David Duncan Ross PalmerASTERISK-24190: IMAP voicemail causes segfault
Reported by: Nick Adams

Category: Applications/app_voicemail/ODBC

ASTERISK-27760: Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version.
Reported by: Nic ColledgeASTERISK-27853: Incorrect error reported when leaving/retrieving a ODBC voicemail
Reported by: Nic ColledgeASTERISK-27093: ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
Reported by: James TerhuneASTERISK-26723: VoiceMailPlayMsg not playing messages via realtime
Reported by: Ryan RittgarnASTERISK-24288: [patch] - ODBC usage with app_voicemail - voicemail is not deleted after review, hangup
Reported by: LEI FU

Category: Bridges/bridge_holding

ASTERISK-25271: Parking & blind transfer: Transferer channel not hung up if no MOH
Reported by: Kevin HarwellASTERISK-24281: When bridging 2 chan_sip channels, MOH not removed from on-hold channels and bridge is never destroyed after hangup.
Reported by: Stefan Engström

Category: Bridges/bridge_native_rtp

ASTERISK-27299: Asterisk Hangs with Bad file descriptor on read()
Reported by: Abhay GuptaASTERISK-27257: bridge_native_rtp: half-way direct media when using early bridging
Reported by: Jean Aunis - PrescomASTERISK-25240: bridge_native_rtp: Direct media wrongfully started when completing attended transfer
Reported by: Joshua C. ColpASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.
Reported by: Rusty NewtonASTERISK-24459: bridge_native_rtp: Native RTP bridging is chosen for RTP compatible channels when the DTMF mode is not compatible
Reported by: Yaniv SimhiASTERISK-24327: bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants
Reported by: Matt JordanASTERISK-24195: bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge
Reported by: Jonathan Rose

Category: Bridges/bridge_simple

ASTERISK-27692: bridging: Sometimes cloning the stream topology causes a crash
Reported by: Richard MudgettASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. VaillancourtASTERISK-26966: bridge_simple: Add support for streams
Reported by: Kevin HarwellASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John Bigelow

Category: Bridges/bridge_softmix

ASTERISK-27939: [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
Reported by: Alexander TraudASTERISK-27804: bridge_softmix / app_confbridge: Add support for combining REMB reports
Reported by: Joshua C. ColpASTERISK-27786: app_confbridge: Add ability to enable and configure REMB support
Reported by: Joshua C. ColpASTERISK-27755: ConfBridge: raise ConfbridgeTalking when put on hold and clear talking status
Reported by: Kevin HarwellASTERISK-27550: [patch] bridge_softmix: Avoid warning about an uninitialized variable.
Reported by: Alexander TraudASTERISK-27354: bridge_softmix: When a channel leaves add in any missing participant streams
Reported by: Joshua C. ColpASTERISK-27277: bridge: Renegotiate if source stream changes.
Reported by: Joshua C. ColpASTERISK-27143: bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
Reported by: Joshua C. ColpASTERISK-27136: bridge_softmix: Don't reorder SFU streams
Reported by: Joshua C. ColpASTERISK-27134: bridge_softmix: Reuse any removed streams for video
Reported by: Joshua C. ColpASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt JordanASTERISK-24797: bridge_softmix: G.729 codec license held
Reported by: Kevin HarwellASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John Bigelow

Category: CDR/General

ASTERISK-27909: cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch
Reported by: Denis LebedevASTERISK-27656: CDR: Leaking channel snapshots allocated by stasis_channel.c
Reported by: Kristijan VrbanASTERISK-27539: 'cdr submit' fails: batch mode not enabled.
Reported by: Tzafrir CohenASTERISK-26818: cdr: Problem setting variables in h exten
Reported by: Sebastian GutierrezASTERISK-26103: cdr: Assert on 'dial end' event during a blond transfer
Reported by: George JosephASTERISK-25458: Unable to set CDR variable in h extension or hangup_handler
Reported by: Ross BeerASTERISK-23904: #define AST_MAX_ACCOUNT_CODE 20 causes truncation
Reported by: Ben MerrillsASTERISK-24344: CDR_PROP(disable) disables CDR only for first dialed party
Reported by: Janusz KarolakASTERISK-24443: CDR fields (dst, dcontext) empty in transfer call started from Macro
Reported by: Arveno SantoroASTERISK-25090: CLI core show channel truncates cdr variables
Reported by: snuffyASTERISK-24426: CDR Batch mode: size used as time value after first expire
Reported by: Shane BlaserASTERISK-24237: CDR: FRACK With PJSIP blonde transfer.
Reported by: Richard MudgettASTERISK-24394: CDR: FRACK with PJSIP directed pickup.
Reported by: Richard MudgettASTERISK-24254: CDRs: Application/args/dialplan CEP updated during dial operation
Reported by: Matt JordanASTERISK-24241: crash: CDRs recursively attempt to update Party B information in a multi-party bridge, overrunning the stack
Reported by: Deepak Singh Rawat

Category: CDR/cdr_adaptive_odbc

ASTERISK-26818: cdr: Problem setting variables in h exten
Reported by: Sebastian GutierrezASTERISK-25263: [patch]cdr_adaptive_odbc: CDR insert failure due to reversed if logic
Reported by: Elazar Broad

Category: CDR/cdr_custom

ASTERISK-27165: CDR: CDR(start,u) function won't work in cdr_custom config
Reported by: Jacek KoniecznyASTERISK-26054: Asterisk crashes (core dump)
Reported by: B. DavisASTERISK-25179: CDR(billsec,f) and CDR(duration,f) report incorrect values
Reported by: Gianluca Merlo

Category: CDR/cdr_odbc

ASTERISK-24976: cdr_odbc not include new columns added on 1.8
Reported by: Rodrigo Ramirez Norambuena

Category: CDR/cdr_pgsql

ASTERISK-24959: [patch]CLI command cdr show pgsql status
Reported by: Rodrigo Ramirez Norambuena

Category: CDR/cdr_radius

ASTERISK-26455: cdr_radius / cel_radius: try fix memory leak
Reported by: Badalian Vyacheslav

Category: CEL/General

ASTERISK-28081: chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces
Reported by: Luit van DrongelenASTERISK-25262: Memory leak when a caller channel does multiple dials and CEL is enabled
Reported by: Etienne LessardASTERISK-25647: bug of cel_radius.c: wrong point of ADD_VENDOR_CODE
Reported by: Aaron AnASTERISK-22367: Rework CEL unit test verification step
Reported by: Kinsey Moore

Category: CEL/cel_odbc

ASTERISK-25032: [patch]cel_odbc sometimes inserts CEL with wrong eventtime
Reported by: Etienne Lessard

Category: CEL/cel_pgsql

ASTERISK-26896: Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
Reported by: twistedASTERISK-24967: Problem support schema for pgsql on CEL
Reported by: Rodrigo Ramirez Norambuena

Category: Channels/General

ASTERISK-27426: chan_console: cannot read and write at the same time with alsa backend
Reported by: Tzafrir CohenASTERISK-27490: chan_console: 'set active' fails to work
Reported by: Tzafrir CohenASTERISK-27289: A codeblock that maintains a bug,but maybe the codeblock will never run
Reported by: HuangyxASTERISK-25025: Periodic crashes (in ast_channel_snapshot_create at stasis_channels.c) with Certified Asterisk 13.
Reported by: Chet StevensASTERISK-24415: Missing AMI VarSet events when channels inherit variables.
Reported by: Richard Mudgett

Category: Channels/chan_alsa

ASTERISK-27720: [patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD.
Reported by: Alexander Traud

Category: Channels/chan_dahdi

ASTERISK-27343: Fails to build in FreeBSD due to sys/sysmacros.h not existing there
Reported by: Guido FalsiASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and missing paren issues
Reported by: George JosephASTERISK-26412: build: Prepare for gcc 6.2
Reported by: George JosephASTERISK-26216: res_fax: Deadlock when detect fax while channel executing Playback
Reported by: Richard MudgettASTERISK-25315: DAHDI channels send shortened duration DTMF tones.
Reported by: Richard MudgettASTERISK-25257: [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope
Reported by: Patric MarschallASTERISK-21893: Segfault after call hangup, in ast_channel_hangupcause_set, at channel_internal_api.c
Reported by: Aleksandr GordeevASTERISK-25034: chan_dahdi: Some telco switches occasionally ignore ISDN RESTART requests.
Reported by: Richard MudgettASTERISK-19608: Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
Reported by: Denis Alberto MartinezASTERISK-24895: After hangup on the side of the ISDN network no HangupRequest event comes for the dahdi channel.
Reported by: Andrew ZherdinASTERISK-24869: Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel
Reported by: viniciusfontesASTERISK-24825: Caller ID not recognized using Centrex/Distinctive dialing
Reported by: Richard MudgettASTERISK-17588: Caller ID on TDM410P *UK* PSTN
Reported by: Daniel FloundersASTERISK-24689: Segfault on hangup after outgoing PRI-Euroisdn call
Reported by: Marcel Manz

Category: Channels/chan_dahdi/NewFeature

ASTERISK-26214: Allow arbitrary time for fax detection to end on a channel
Reported by: Richard Mudgett

Category: Channels/chan_h323

ASTERISK-27670: [patch] BuildSystem: Remove chan_h323 leftovers.
Reported by: Alexander Traud

Category: Channels/chan_iax2

ASTERISK-27705: chan_iax2: Stops listening for traffic
Reported by: Kirsty TyermanASTERISK-27908: [patch] crypto.h: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-27122: chan_iax2: On reload MWI taskprocessors keep adding up
Reported by: Sergej KasumovicASTERISK-26865: chan_iax2: Reload of iax peer results in loss of host address/port
Reported by: Richard BeggASTERISK-22820: [patch] Plaintext auth is still supported in IAX2
Reported by: EugeneASTERISK-24983: IAX deadlock between hangup and scheduled actions (ex. largrq)
Reported by: Y AteyaASTERISK-22352: [patch] IAX2 custom qualify timer is not taken into account
Reported by: Frederic Van EspenASTERISK-24894: [patch] iax2_poke_noanswer expiration timer too short
Reported by: Y AteyaASTERISK-21211: chan_iax2 - unprotected access of iaxs[peer->callno] potentially results in segfault
Reported by: Jaco KroonASTERISK-24451: chan_iax2: reference leak in sched_delay_remove
Reported by: Corey FarrellASTERISK-24600: Stuck IAX channels, Asterisk stops responding to most traffic, potential deadlock
Reported by: Jeff CollellASTERISK-24389: chan_iax2: Unit test on Bamboo failing
Reported by: Kevin HarwellASTERISK-24265: segfault in asterisk when try to make call to IAX
Reported by: Dafi NiASTERISK-23767: [patch] Dynamic IAX2 registration stops trying if ever not able to resolve
Reported by: David Herselman

Category: Channels/chan_local

ASTERISK-25649: Transfer application does not work with Local channels - documentation misleading
Reported by: Ivan UllmannASTERISK-25912: chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding them to the local hangupcauses via ast_channel_hangupcause_hash_set
Reported by: Jaco KroonASTERISK-25250: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback
Reported by: Etienne LessardASTERISK-24267: Queue variables associated with setinterfacevar, setqueueentryvar, setqueuevar are not passed to local channel
Reported by: Mitch ClabornASTERISK-22409: Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade
Reported by: Corey FarrellASTERISK-24415: Missing AMI VarSet events when channels inherit variables.
Reported by: Richard Mudgett

Category: Channels/chan_mgcp

ASTERISK-25220: [patch]Closing of fd -1 in chan_mgcp.c
Reported by: Walter DoekesASTERISK-24500: Regression introduced in chan_mgcp by SVN revision r227276
Reported by: Xavier Hienne

Category: Channels/chan_motif

ASTERISK-24384: chan_motif: format capabilities leak on module load error
Reported by: Corey Farrell

Category: Channels/chan_multicast_rtp

ASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode

Category: Channels/chan_phone

ASTERISK-24458: chan_phone fails to build on big endian systems
Reported by: Tzafrir Cohen

Category: Channels/chan_pjsip

ASTERISK-28538: chan_pjsip: Deadlock on fax detection
Reported by: Joshua C. ColpASTERISK-28322: chan_pjsip: Add option to allow ignoring of 183 without SDP
Reported by: Torrey SearleASTERISK-28213: res_pjsip: Threads pile up needlessly when AOR is blocked
Reported by: Ross BeerASTERISK-28238: PJSIP realtime. getcontext not working with DUNDI
Reported by: RayASTERISK-27095: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE
Reported by: George JosephASTERISK-27999: Wrong SRTP use status report
Reported by: Salah AhmedASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses
Reported by: George JosephASTERISK-27554: res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints
Reported by: Ross BeerASTERISK-27441: PJSIP: Forked INVITE SDP negotiation gets one way audio.
Reported by: lvlASTERISK-27568: PJSIP: Crash during SIP attended transfer.
Reported by: Bryan WaltersASTERISK-27612: Subscriptions Persist After Expiration and TCP/TLS Disconnect
Reported by: Ross BeerASTERISK-26832: res_pjsip: Segfault when calling pjsip_hdr_print_on in sip_msg.c:581
Reported by: Ross BeerASTERISK-27480: Security: Authenticated SUBSCRIBE without Contact crashes asterisk
Reported by: Ross BeerASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel
Reported by: Zane ConkleASTERISK-27259: chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
Reported by: lvlASTERISK-27248: [patch]external_media_address and external_signaling_address don't always honor localnet
Reported by: Walter DoekesASTERISK-27236: Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
Reported by: Ross BeerASTERISK-27076: chan_pjsip: Add support for multiple streams
Reported by: Joshua C. ColpASTERISK-27039: chan_pjsip: Device state is idle when channel from endpoint is in early media
Reported by: Joshua C. ColpASTERISK-26996: chan_pjsip: Flipping between codecs
Reported by: Michael MaierASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints
Reported by: Jacek KoniecznyASTERISK-26857: chan_pjsip: Dialplan function race condition
Reported by: Joshua C. ColpASTERISK-26822: pjsip/cli_commands: pjsip show channelstats shows wrong codec
Reported by: Kevin HarwellASTERISK-26248: chan_pjsip: Error when calling PJSIP client with domain specified
Reported by: Norbert VargaASTERISK-26673: chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
Reported by: Joshua C. ColpASTERISK-26603: [patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
Reported by: Alexei GradinariASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard MudgettASTERISK-26482: [patch] chan_pjsip: segfault on already disconnected session
Reported by: Alexei GradinariASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John KinistonASTERISK-26396: chan_pjsip: HANGUPCAUSE return the wrong code when dialed channel answer.
Reported by: Aaron AnASTERISK-26306: channel: Hang-up crashes, chan_pjsip not cleaning up properly
Reported by: Alexander TraudASTERISK-26145: pjsip: Deadlock with suspend + masquerade + indicate
Reported by: Ross BeerASTERISK-26216: res_fax: Deadlock when detect fax while channel executing Playback
Reported by: Richard MudgettASTERISK-26214: Allow arbitrary time for fax detection to end on a channel
Reported by: Richard MudgettASTERISK-26063: ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
Reported by: Private NameASTERISK-24986: keepalive INFO packages ignored by asterisk
Reported by: Ilya TrikozASTERISK-26005: res_pjsip: Multiple SIP messages are combined into 1 TCP packet
Reported by: Ross BeerASTERISK-25990: PJSIP TLS registration should respect client_uri scheme when generating Contact URI
Reported by: Sebastian DammASTERISK-25826: PJSIP / Sorcery slow load from realtime
Reported by: Ross BeerASTERISK-25849: chan_pjsip: transfers with direct media sometimes drops audio
Reported by: Kevin HarwellASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic ColledgeASTERISK-25637: Multi homed server using wrong IP
Reported by: Daniel JournoASTERISK-25675: Endpoint not listed as Unreachable
Reported by: Daniel JournoASTERISK-24779: Passthrough OPUS codec not working with chan_pjsip
Reported by: PowerPBXASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql
Reported by: mdu113ASTERISK-25404: segfault/crash in chan_pjsip_hangup ... at chan_pjsip.c
Reported by: Chet StevensASTERISK-25258: chan_pjsip: Incorrect format switch on received RTP packet
Reported by: Joshua C. ColpASTERISK-25183: PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite previous checks for NULL channel
Reported by: Matt JordanASTERISK-25091: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge
Reported by: Ilya TrikozASTERISK-25156: chan_pjsip’s CHAN_START cel event lacks the correct context and exten
Reported by: cloosASTERISK-24996: chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf
Reported by: Ashley SandersASTERISK-25018: pjsip show endpoints crashes asterisk when qualified aors present
Reported by: Ivan PoddubnyASTERISK-24845: pjsip send notify not working with Cisco phone
Reported by: Carl FortinASTERISK-24933: T38 fails negotiation
Reported by: Jonathan RoseASTERISK-24781: PJSIP: Unnecessary 180 Ringing messages sent with undesireabe consequences.
Reported by: Richard MudgettASTERISK-24771: ${CHANNEL(pjsip)} - segfault
Reported by: Niklas LarssonASTERISK-24666: Security Vulnerability: RTP not closed after sip call using unsupported codec
Reported by: Y AteyaASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
Reported by: Niklas LarssonASTERISK-24556: Asterisk 13 core dumps when calling from pjsip extension to another pjsip extension
Reported by: Abhay GuptaASTERISK-24382: chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK
Reported by: Matt JordanASTERISK-24356: PJSIP: Directed pickup causes deadlock
Reported by: Richard MudgettASTERISK-24222: PJSIP: Failed assertions when placing a call with no allow= specified
Reported by: Mark MichelsonASTERISK-24271: Unable to make WebRTC call through chan_PJSIP nor chan_SIP
Reported by: Dafi NiASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Reported by: Matt JordanASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
Reported by: Aleksei Kulakov

Category: Channels/chan_rtp

ASTERISK-26672: Crash when setting remote address on RTP instance
Reported by: Richard Mudgett

Category: Channels/chan_sip/CodecHandling

ASTERISK-26691: Remember SDP negotiation on SIP_CODEC_INBOUND.
Reported by: Alexander TraudASTERISK-24543: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
Reported by: Taylor HawkesASTERISK-25160: [patch] Opus Codec: SIP/SDP line fmtp missing when called internally
Reported by: Alexander TraudASTERISK-25484: [patch] autoframing=yes has no effect
Reported by: Alexander TraudASTERISK-17410: Video dynamic RTP payload type negotiation incorrect when directmedia enabled
Reported by: Boris FoxASTERISK-25309: [patch] iLBC 20 advertised
Reported by: Alexander TraudASTERISK-25182: [patch] on CLI sip reload, new codecs get appended only
Reported by: Alexander TraudASTERISK-21777: Asterisk tries to transcode video instead of audio
Reported by: Nick Ruggles

Category: Channels/chan_sip/DatabaseSupport

ASTERISK-25934: chan_sip should not require sipregs or updateable sippeers table unless rt
Reported by: Jaco KroonASTERISK-24772: ODBC error in realtime sippeers when device unregisters under MariaDB
Reported by: Richard Miller

Category: Channels/chan_sip/General

ASTERISK-28362: strtok_r() makes gcc compile warning
Reported by: sungtae kimASTERISK-25792: chan_sip: qualifygap bounds checking
Reported by: Paul SandysASTERISK-28194: chan_sip: Leak using contact ACL
Reported by: Giuseppe SucameliASTERISK-28081: chan_sip: Asterisk 12+ chan_sip doesn't report AST_CEL_PICKUP in handle_invite_replaces
Reported by: Luit van DrongelenASTERISK-27674: chan_sip: RTP framing issues on outgoing calls
Reported by: Jean Aunis - PrescomASTERISK-24488: Wrong remote identity and target in dialog package XML in NOTIFY
Reported by: Alejandro PadillaASTERISK-27646: ICE fails with no candidate nominated
Reported by: Thomas GuebelsASTERISK-27666: chan_sip: Crash processing CANCEL request
Reported by: Leandro DardiniASTERISK-27534: chan_sip: Assumes iostream is non-NULL when it may not be
Reported by: Lubos DolezelASTERISK-27498: ICE candidate parser - ICE foundation parsing too short
Reported by: Michele PràASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel
Reported by: Zane ConkleASTERISK-26131: chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call to a single character in a dot pattern match
Reported by: Dwayne HubbardASTERISK-20643: SIP ICE support - remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message
Reported by: RoyASTERISK-27412: core: Audiohook freeing interpolated frame when it shouldn't.
Reported by: MikhailASTERISK-23462: Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't
Reported by: Rusty NewtonASTERISK-26922: chan_sip: tcpbind uses wrong source address
Reported by: KseniaASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
Reported by: Alexander TraudASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngströmASTERISK-26951: chan_sip: ACK with SDP does not update a direct media bridge
Reported by: Jean Aunis - PrescomASTERISK-26692: res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
Reported by: Sebastian GutierrezASTERISK-26897: chan_sip: Security vulnerability with client code header
Reported by: Alex Villacís LassoASTERISK-26841: chan_sip: Call not cancelled after receiving a 422 response
Reported by: Jean Aunis - PrescomASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and missing paren issues
Reported by: George JosephASTERISK-26573: Some typos in documentation of chan_sip.c
Reported by: C.J. CollierASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
Reported by: Michael KeuterASTERISK-26476: chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
Reported by: Sergey GrachevASTERISK-26457: [patch] force_rport,auto_comedia: No NAT detection triggered.
Reported by: Alexander TraudASTERISK-25468: Deadlock in chan_sip - core show locks shows do_monitor lock
Reported by: Barry FlanaganASTERISK-26358: chan_sip: Contact is updated on re-200, but not on re-INVITE
Reported by: Walter DoekesASTERISK-26272: chan_sip: File descriptors leak (UDP sockets)
Reported by: Etienne LessardASTERISK-24822: Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
Reported by: David BrillertASTERISK-23013: [patch] Deadlock between 'sip show channels' command and attended transfer handling
Reported by: Ben SmithurstASTERISK-26216: res_fax: Deadlock when detect fax while channel executing Playback
Reported by: Richard MudgettASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code.
Reported by: Corey FarrellASTERISK-26193: chan_sip: reference leak in mwi_event_cb
Reported by: Corey FarrellASTERISK-26184: chan_sip: Reference leaks in error paths.
Reported by: Corey FarrellASTERISK-26069: Asterisk truncates To: header, dropping the closing '>'
Reported by: Vasil KolevASTERISK-25950: [patch]SIP channel does not send PeerStatus events for autocreated peers
Reported by: Kirill KatsnelsonASTERISK-25927: Removed option "registertrying" is still documented in sip.conf.sample
Reported by: Etienne LessardASTERISK-24543: Asterisk 13 responds to SIP Invite with all possible codecs configured for peer as opposed to intersection of configured codecs and offered codecs
Reported by: Taylor HawkesASTERISK-21301: ERROR and failure to resolve socket address due to whitespace after port number in SIP Via header
Reported by: Martin VitASTERISK-25023: Deadlock in chan_sip in update_provisional_keepalive
Reported by: Arnd SchmitterASTERISK-25397: [patch]chan_sip: File descriptor leak with non-default timert1
Reported by: Alexander TraudASTERISK-25364: [patch]Issue a TCP connection(kernel) and thread of asterisk is not released
Reported by: Hiroaki KomatsuASTERISK-25610: Asterisk crash during "sip reload"
Reported by: Dudás JózsefASTERISK-25476: chan_sip loses registrations after a while
Reported by: Michael KeuterASTERISK-25346: chan_sip: Overwriting answered elsewhere hangup cause on call pickup
Reported by: Joshua C. ColpASTERISK-25250: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback
Reported by: Etienne LessardASTERISK-22805: res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
Reported by: Dmitry BurilovASTERISK-25212: [patch]Segfault when using DEBUG_FD_LEAKS
Reported by: Walter DoekesASTERISK-25202: Hints extension state broken between 13.3.2 and 13.4
Reported by: Marek CervenkaASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.
Reported by: Rusty NewtonASTERISK-25163: Deadlock in chan_sip between reload of sip peer container and MWI Stasis callback
Reported by: Dmitriy SerovASTERISK-24835: Early Media Not working with Chan SIP and Asterisk 13
Reported by: Andrew NagyASTERISK-24882: chan_sip: Improve usage of REF_DEBUG
Reported by: Corey FarrellASTERISK-24876: Investigate reference leaks from tests/channels/local/local_optimize_away
Reported by: Corey FarrellASTERISK-24838: chan_sip: Locking inversion occurs when building a peer causes a peer poke during request handling
Reported by: Richard MudgettASTERISK-21845: maxcalls exceeded, Asterisk sends out 480 and also BYE
Reported by: Tony ChingASTERISK-15434: [patch] When ast_pbx_start failed, both an error response and BYE are sent to the caller
Reported by: Makoto DeiASTERISK-23214: chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases
Reported by: Rusty NewtonASTERISK-24800: Crash in __sip_reliable_xmit due to invalid thread ID being passed to pthread_kill
Reported by: JoshEASTERISK-24355: [patch] chan_sip realtime uses case sensitive column comparison for 'defaultuser'
Reported by: HZMI8gkCvPpom0tMASTERISK-24628: [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
Reported by: Karsten WemheuerASTERISK-24533: 2 threads created per chan_sip entry
Reported by: xrobauASTERISK-24281: When bridging 2 chan_sip channels, MOH not removed from on-hold channels and bridge is never destroyed after hangup.
Reported by: Stefan EngströmASTERISK-24307: Unintentional memory retention in stringfields
Reported by: Etienne LessardASTERISK-24063: [patch]Asterisk does not respect outbound proxy when sending qualify requests
Reported by: Damian IvereighASTERISK-24321: SIP deadlock when running automated queues tests
Reported by: Steve PittsASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE
Reported by: not hereASTERISK-20784: Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
Reported by: NITESH BANSALASTERISK-15879: [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
Reported by: Torrey SearleASTERISK-22945: [patch] Memory leaks in chan_sip.c with realtime peers
Reported by: ibercomASTERISK-24335: [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls
Reported by: Torrey SearleASTERISK-24271: Unable to make WebRTC call through chan_PJSIP nor chan_SIP
Reported by: Dafi NiASTERISK-24178: [patch]fromdomainport used even if not set
Reported by: Elazar Broad

Category: Channels/chan_sip/IPv6

ASTERISK-27434: [patch] chan_sip/ICE: Square brackets around IPv6 addresses.
Reported by: Alexander TraudASTERISK-26438: [patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
Reported by: Alexander TraudASTERISK-18232: Broken REGISTER sent to IPv4 server when bindaddr=[::]
Reported by: JacekASTERISK-25443: [patch]IPv6 - Potential issue in via header parsing
Reported by: ffsASTERISK-25100: asterisk coredump if host has an IPv6 address that end with ::80
Reported by: Mark PetersenASTERISK-18032: [patch] - IPv6 and IPv4 NAT not working
Reported by: Christoph Timm

Category: Channels/chan_sip/Interoperability

ASTERISK-18140: Expires handling in SUBSCRIBE confuses the absence of the Expires header field with an unsubscribe action.
Reported by: Jonathan ClootsASTERISK-27365: [patch] chan_sip: Crypto attribute not last but first on SDP media level.
Reported by: Alexander TraudASTERISK-17540: SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
Reported by: saghulASTERISK-21721: SIP Failed to parse multiple Supported: headers
Reported by: Olle JohanssonASTERISK-26915: chan_sip: Session Timers required but refused wrongly.
Reported by: Alexander TraudASTERISK-26433: chan_sip: Allows To-tag checks to be bypassed, setting up new calls
Reported by: Walter DoekesASTERISK-26030: call cut because of double Session-Expires header in re-invite after proxy authentication is required
Reported by: George JosephASTERISK-25135: [patch]RTP Timeout hangup cause code missing
Reported by: Olle JohanssonASTERISK-25396: chan_sip: Extremely long callerid name causes invalid SIP
Reported by: Walter DoekesASTERISK-25154: [patch]fromtag may need to be updated after successful call dialog match
Reported by: Damian IvereighASTERISK-24646: PJSIP changeset 4899 breaks TLS
Reported by: Stephan Eisvogel

Category: Channels/chan_sip/Messaging

ASTERISK-28057: chan_sip: SipNotify via AMI behaves differently to CLI
Reported by: Peter KatzmannASTERISK-24301: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
Reported by: Matt Jordan

Category: Channels/chan_sip/NewFeature

ASTERISK-24280: Add 'rtpbindaddr' setting for chan_sip
Reported by: Paul Belanger

Category: Channels/chan_sip/Registration

ASTERISK-18232: Broken REGISTER sent to IPv4 server when bindaddr=[::]
Reported by: JacekASTERISK-25950: [patch]SIP channel does not send PeerStatus events for autocreated peers
Reported by: Kirill KatsnelsonASTERISK-24715: chan_sip: stale nonce causes failure
Reported by: Kevin HarwellASTERISK-24673: outgoing sip registers cannot be removed or modified without doing restart (or doing module unload chan_sip.so)
Reported by: Stefan EngströmASTERISK-24640: Registration pending stays forever after sip reload
Reported by: Max Man

Category: Channels/chan_sip/SRTP

ASTERISK-27795: chan_sip: one way / no audio with srtp
Reported by: Florian KaiserASTERISK-27395: srtp: Add support for ephemeral DTLS certificates
Reported by: Sean BrightASTERISK-27365: [patch] chan_sip: Crypto attribute not last but first on SDP media level.
Reported by: Alexander TraudASTERISK-16898: SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0
Reported by: Marcello CeschiaASTERISK-25490: [patch]SDP crypto tag is validated incorrectly
Reported by: Joerg SonnenbergerASTERISK-20234: SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
Reported by: tootaiASTERISK-23989: [patch]SDP offer/answer fails if crypto keys added to non-crypto offer
Reported by: Olle JohanssonASTERISK-24550: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
Reported by: Osaulenko AlexanderASTERISK-24887: [patch]tags in a=crypto lines do not accept 2 or more digits
Reported by: Makoto DeiASTERISK-17721: Incoming SRTP calls that specify a key lifetime fail
Reported by: Terry WilsonASTERISK-20233: SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
Reported by: tootaiASTERISK-22748: SRTP Crypto Offer With Lifetime Not Accepted
Reported by: Alejandro Mejia

Category: Channels/chan_sip/Security Framework

ASTERISK-25869: chan_sip: "rejected because extension not found" should be logged as a security event
Reported by: Brian J. MurrellASTERISK-25722: ASAN & testsute: stack-buffer-overflow in sip_sipredirect
Reported by: Badalian VyacheslavASTERISK-25320: chan_sip.c: sip_report_security_event searches for wrong or non existent peer on invite
Reported by: Kevin Harwell

Category: Channels/chan_sip/Subscriptions

ASTERISK-28173: Deadlock in chan_sip handling subscribe request during res_parking reload
Reported by: Giuseppe SucameliASTERISK-27217: chan_sip: Asterisk crashing when subscription doesn't get set
Reported by: Bryan Walters

Category: Channels/chan_sip/T.38

ASTERISK-26179: chan_sip: Second T.38 request fails
Reported by: Joshua C. ColpASTERISK-25609: [patch]Asterisk may crash when calling ast_channel_get_t38_state(c)
Reported by: Filip JenicekASTERISK-24449: Reinvite for T.38 UDPTL fails if SRTP is enabled
Reported by: Andreas SteinmetzASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE
Reported by: not here

Category: Channels/chan_sip/TCP-TLS

ASTERISK-28057: chan_sip: SipNotify via AMI behaves differently to CLI
Reported by: Peter KatzmannASTERISK-28034: chan_sip unstable with TLS after asterisk start or reloads
Reported by: David HajekASTERISK-27881: PBX calls via chan_sip TCP trunk now get authentification error
Reported by: Ian GilmourASTERISK-27457: chan_sip: Guests disallowed via TCP (or TLS) if existing peer from same IP.
Reported by: Alexander TraudASTERISK-27339: [patch] Crash on ast_ssl_teardown when stopping.
Reported by: Alexander TraudASTERISK-27324: [patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
Reported by: Alexander TraudASTERISK-26586: chan_sip: Segfaults upon reload if client with MWI wasn't registered
Reported by: Michael KuronASTERISK-26604: chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
Reported by: Michael KuronASTERISK-19968: TCP Session-Timers not dropping call
Reported by: Aaron HamstraASTERISK-24847: [security] [patch] tcptls: certificate CN NULL byte prefix bug
Reported by: Matt JordanASTERISK-22748: SRTP Crypto Offer With Lifetime Not Accepted
Reported by: Alejandro MejiaASTERISK-24799: [patch] make fails with undefined reference to SSLv3_client_method
Reported by: Alexander Traud

Category: Channels/chan_sip/Transfers

ASTERISK-27740: chan_sip: New Channel creation from new SIP dialog with Replaces failed to be properly tracked and destroyed
Reported by: Shannon PriceASTERISK-25226: chan_sip: Channel leak in branch 13 on early replaces call pickup
Reported by: Walter DoekesASTERISK-24628: [patch] chan_sip - CANCEL is sent to wrong destination when 'sendrpid=yes' (in proxy environment)
Reported by: Karsten WemheuerASTERISK-15242: transmit_refer leaks sip_refer structures
Reported by: David Woolley

Category: Channels/chan_sip/Video

ASTERISK-17470: [patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
Reported by: effie mouzeli

Category: Channels/chan_sip/WebSocket

ASTERISK-24330: Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
Reported by: Marek CervenkaASTERISK-24146: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
Reported by: Aleksei KulakovASTERISK-23997: chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer
Reported by: Badalian Vyacheslav

Category: Channels/chan_skinny

ASTERISK-27452: Security: chan_skinny: Memory exhaustion if flooded with unauthenticated requests
Reported by: George JosephASTERISK-26940: Asterisk Skinny memory exhaustion vulnerability leads to DoS
Reported by: Sandro GauciASTERISK-25494: build: GCC 5.1.x catches some new const, array bounds and missing paren issues
Reported by: George JosephASTERISK-26029: parking: ast_parking_park_call should return parking_space instead of parking_exten
Reported by: Diederik de GrootASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard Mudgett

Category: Channels/chan_unistim

ASTERISK-27714: [patch] chan_unistim: NetBSD has an incompatible struct in_pktinfo.
Reported by: Alexander TraudASTERISK-26596: Placing call on hold temporarily locks up set
Reported by: Igor GoncharovskyASTERISK-26714: Phone default have not ringing on ARM
Reported by: Igor GoncharovskyASTERISK-26565: chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
Reported by: RuseASTERISK-26138: chan_unistim: Under FreeBSD, chan_unistim generates a compile error
Reported by: George JosephASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard MudgettASTERISK-24304: asterisk crashing randomly because of unistim channel
Reported by: dhanapathy sathyaASTERISK-23846: Unistim multilines. Loss of voice after second call drops (on a second line).
Reported by: Rustam Khankishyiev

Category: Channels/chan_vpb

ASTERISK-27808: [patch] chan_vpb: Avoid GNU old-style field designator extension.
Reported by: Alexander Traud

Category: Codecs/General

ASTERISK-27814: translate: interpolated frames are not passed through
Reported by: Kevin HarwellASTERISK-23735: Transcoding makes bad choice in high-rate translations
Reported by: Richard KennerASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
Reported by: Frankie ChinASTERISK-26812: [patch] Fix download_externals To Allow The Use Of curl Or wget
Reported by: Michael L. YoungASTERISK-26144: Crash on loading codecs g729/g723
Reported by: Alexei GradinariASTERISK-25914: PJSIP: failed registration with wrong codec name on allow/disallow
Reported by: Alexei GradinariASTERISK-25616: Warning with a Codec Module which supports PLC with FEC
Reported by: Alexander TraudASTERISK-25498: Asterisk crashes when negotiating g729 without that module installed
Reported by: Ben LangfeldASTERISK-25353: [patch] Transcoding while different in Frame size = Frames lost
Reported by: Alexander Traud

Category: Codecs/codec_adpcm

ASTERISK-24717: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
Reported by: Badalian Vyacheslav

Category: Codecs/codec_dahdi

ASTERISK-24435: Asterisk 13 with TC400P segfault
Reported by: Marian Koniuszko

Category: Codecs/codec_g722

ASTERISK-27232: When in queue on g722 with interruptions, music on hold can get stuck and no longer play
Reported by: Jens T.

Category: Codecs/codec_gsm

ASTERISK-27558: [patch] codec_gsm: Avoid shifting a negative signed value.
Reported by: Alexander TraudASTERISK-24717: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
Reported by: Badalian Vyacheslav

Category: Codecs/codec_ilbc

ASTERISK-27669: [patch] codecs: Add support for WebRTC iLBC 2.0.
Reported by: Alexander TraudASTERISK-24717: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
Reported by: Badalian Vyacheslav

Category: Codecs/codec_lpc10

ASTERISK-24717: ASAN: global-buffer-overflow codec_{ilbc | gsm | adpcm | ipc10}
Reported by: Badalian Vyacheslav

Category: Codecs/codec_opus

ASTERISK-28263: codec_opus: errors setting max_playback_rate and bitrate to "sdp"
Reported by: Gianluca MerloASTERISK-27202: If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
Reported by: Seán C. McCordASTERISK-26520: codec_opus: Generated fmtp line has no content
Reported by: Sebastian Gutierrez

Category: Codecs/codec_resample

ASTERISK-25599: [patch] SLIN Resampling Codec only 80 msec
Reported by: Alexander Traud

Category: Codecs/codec_siren14

ASTERISK-16172: Problems with siren14 codec; problems with siren7 sound files.
Reported by: Steve MurphyASTERISK-26021: Build codecs siren7 and siren14 for Asterisk 13
Reported by: Daniel Denson

Category: Codecs/codec_siren7

ASTERISK-16172: Problems with siren14 codec; problems with siren7 sound files.
Reported by: Steve MurphyASTERISK-27202: If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
Reported by: Seán C. McCordASTERISK-26021: Build codecs siren7 and siren14 for Asterisk 13
Reported by: Daniel Denson

Category: Configs/Basic-PBX

ASTERISK-28272: The basic-pbx config samples don't produce a running asterisk
Reported by: George Joseph

Category: Configs/Samples

ASTERISK-27175: iax.conf demo peer is invalid
Reported by: Tzafrir CohenASTERISK-26785: configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
Reported by: Torrey Searle

Category: Contrib/General

ASTERISK-28323: pjsip: sip.conf to pjsip.conf conversion script fails
Reported by: Guido WeckwerthASTERISK-27968: systemd: asterisk.service
Reported by: seanchann.zhouASTERISK-27811: [patch] sip_to_pjsip: Enable python3 compatibility.
Reported by: Alexander TraudASTERISK-27684: [patch] install_prereq: Update OpenBSD libraries.
Reported by: Alexander TraudASTERISK-27555: [patch] install_prereq: Update Debian/Ubuntu libraries.
Reported by: Alexander TraudASTERISK-27599: [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
Reported by: Alexander TraudASTERISK-27603: [patch] install_prereq: Download latest Jansson.
Reported by: Alexander TraudASTERISK-27598: [patch] install_prereq: Support package manager DNF.
Reported by: Alexander TraudASTERISK-27333: sip_to_pjsip not correctly handling disallow=all directive
Reported by: Torrey SearleASTERISK-24311: Populating database via Alembic fails when using same database for multiple schema sets
Reported by: Dafi NiASTERISK-22374: Finish mapping the sip.conf parameters to res_sip.conf parameters
Reported by: Matt JordanASTERISK-26183: alembic: error when using sqlalchemy version 1.1.0b2
Reported by: Kevin HarwellASTERISK-26128: Alembic scripts are failing
Reported by: Mark MichelsonASTERISK-25890: Asterisk 13.8.0 alembic database update fails
Reported by: Harley PetersASTERISK-25113: install_prereq in Debian 8 without "standard system utilities"
Reported by: Rodrigo Ramirez NorambuenaASTERISK-24632: install_prereq script installs pjproject without IPv6 support
Reported by: Rusty NewtonASTERISK-24048: [patch] contrib/scripts/install_prereq selects 32-bit packages on 64-bit hosts
Reported by: Ben KlangASTERISK-24474: sip_to_pjsip.py lacks documentation and does not function
Reported by: John KinistonASTERISK-24432: Install refcounter.py when REF_DEBUG is enabled
Reported by: Corey FarrellASTERISK-24011: [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM
Reported by: Michael MylesASTERISK-23781: outgoing missing as enum from contrib/ast-db-manage/config
Reported by: Stephen More

Category: Core/ACL

ASTERISK-24969: Named ACL's do not handle config errors.
Reported by: Corey Farrell

Category: Core/AstDB

ASTERISK-27706: PJSIP: Deadlock shutting down subscription TCP connection and sending subscription message.
Reported by: Ross BeerASTERISK-25400: Hints broken when "CustomPresence" doesn't exist in AstDB
Reported by: Andrew Nagy

Category: Core/AstMM

ASTERISK-26526: [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
Reported by: Badalian VyacheslavASTERISK-26524: astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
Reported by: Corey FarrellASTERISK-26181: REF_DEBUG: Node object incorrectly logged during duplicate replacement
Reported by: Corey FarrellASTERISK-25120: Astobj2: Weakproxy subscriptions should be run in reverse order.
Reported by: Corey FarrellASTERISK-25048: Astobj2: Initialization order wrong when both refdebug and AO2_DEBUG are both enabled.
Reported by: Corey FarrellASTERISK-24936: New Feature: AO2 weakproxy objects
Reported by: Corey FarrellASTERISK-24535: stringfields: Fix regression from fix for unintentional memory retention and another issue exposed by the fix
Reported by: Corey FarrellASTERISK-24307: Unintentional memory retention in stringfields
Reported by: Etienne Lessard

Category: Core/Bridging

ASTERISK-28076: bridging: Asterisk crashes when receiving an empty realtime text frame
Reported by: Emmanuel BUUASTERISK-27229: bridge: Old channel video source not set to NULL after unref
Reported by: Richard KennerASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel
Reported by: Zane ConkleASTERISK-27238: Bridging: Crash freeing a frame that's already been freed
Reported by: Richard KennerASTERISK-27369: Bridge() dialplan application fails without setting BRIDGERESULT channel variable
Reported by: James TerhuneASTERISK-27182: bridge: Crash when mapping streams
Reported by: Joshua C. ColpASTERISK-27075: bridge: stuck channel(s) after failed attended transfer
Reported by: Kevin HarwellASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
Reported by: Chris HowardASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge
Reported by: Torrey SearleASTERISK-24529: Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
Reported by: Corey FarrellASTERISK-26880: Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
Reported by: Kirsty TyermanASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt JordanASTERISK-25947: Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object.
Reported by: Richard MudgettASTERISK-24782: StasisEnd event not present for channel that was swapped out for another after completing attended transfer
Reported by: John BigelowASTERISK-25771: ARI:Crash - Attended transfers of channels into Stasis application.
Reported by: Javier Riveros ASTERISK-25600: bridging: Inconsistency in BRIDGEPEER
Reported by: Jonathan RoseASTERISK-25341: bridge: Hangups may get lost when executing actions
Reported by: Joshua C. ColpASTERISK-25250: chan_sip - Despite the channel being answered, caller on a call established via Local channel continues to hear ringback
Reported by: Etienne LessardASTERISK-25157: bridging: Performing a blonde transfer does not result in connected line updates
Reported by: Joshua C. ColpASTERISK-24869: Asterisk segfaults on DAHDI attended transfer due to application (appl) being NULL on unbridged channel
Reported by: viniciusfontesASTERISK-24752: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
Reported by: Richard MudgettASTERISK-24539: Compile fails on OSX because of sem_timedwait in bridge_channel.c
Reported by: George JosephASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John BigelowASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
Reported by: Niklas LarssonASTERISK-24447: Bridge DTMF hooks: Audio doesn't pass when waiting for more matching digits.
Reported by: Richard MudgettASTERISK-24437: Review implementation of ast_bridge_impart for leaks and document proper usage
Reported by: Scott Griepentrog

Category: Core/Bridging/bridge_basic

ASTERISK-27361: Attended transfer crashes in Asterisk 13.17.2
Reported by: Alessandro PimentaASTERISK-25696: bridge_basic: don't cache xferfailsound during a transfer
Reported by: Kevin HarwellASTERISK-25697: bridge_basic: don't play an attended transfer fail sound after target hangs up
Reported by: Kevin HarwellASTERISK-25641: bridge: GOTO_ON_BLINDXFR doesn't work on transfer initiated channel
Reported by: Dmitry MelekhovASTERISK-24513: Local channel apparently leaked in off-nominal DTMF attended transfer
Reported by: Mark Michelson

Category: Core/BuildSystem

ASTERISK-28271: Opensuse Leap 15 --with-jannson-bundled will not compile
Reported by: David WilcoxASTERISK-28250: build: Cross-compilation fails for target arm-linux-gnueabihf
Reported by: Jean Aunis - PrescomASTERISK-27991: BuildSystem: Enable Jansson in Solaris 11.
Reported by: Alexander TraudASTERISK-27563: pjsip modules always get -O2 even when DONT_OPTIMIZE is set
Reported by: George JosephASTERISK-27931: [patch] BuildSystem: Enable ./configure in Solaris 11.
Reported by: Alexander TraudASTERISK-27926: [patch] bootstrap.sh: find -maxdepth is not POSIX compatible.
Reported by: Alexander TraudASTERISK-27903: menuselect: GCC 8: restrict-qualified parameter passed and aliased.
Reported by: Alexander TraudASTERISK-27824: Fix issues exposed by GCC 8
Reported by: George JosephASTERISK-27435: [patch] configure: pjsip_evsub_set_uas_timeout not found.
Reported by: Alexander TraudASTERISK-27761: [patch] BuildSystem: With external editline, do not require libs for internal editline.
Reported by: Alexander TraudASTERISK-27745: [patch] BuildSystem: Remove unused dependency on libltdl.
Reported by: Alexander TraudASTERISK-27720: [patch] BuildSystem: Enable Advanced Linux Sound Architecture (ALSA) in NetBSD.
Reported by: Alexander TraudASTERISK-27734: [patch] BuildSystem: Enable IMAP storage on openSUSE and Arch Linux.
Reported by: Alexander TraudASTERISK-27686: [patch] install_prereq: Update FreeBSD libraries.
Reported by: Alexander TraudASTERISK-11015: NetBSD Build Needs RPATH set in 1.2.25
Reported by: Curt SampsonASTERISK-27641: BuildSystem: Enable Better Backtraces in FreeBSD.
Reported by: Alexander TraudASTERISK-25586: uuid_generate_random detection failure
Reported by: John NemethASTERISK-27721: [patch] BuildSystem: Enable PortAudio in NetBSD.
Reported by: Alexander TraudASTERISK-27715: [patch] BuildSystem: AC_PATH_PROG sets to colon character when not found.
Reported by: Alexander TraudASTERISK-27718: [patch] BuildSystem: Enable Lua in NetBSD.
Reported by: Alexander TraudASTERISK-27722: [patch] BuildSystem: Depend not implicitly but explicitly on external libraries.
Reported by: Alexander TraudASTERISK-27716: [patch] BuildSystem: Enable autotools in NetBSD.
Reported by: Alexander TraudASTERISK-27713: [patch] BuildSystem: Cast any intptr_t explicitly to its proposed type.
Reported by: Alexander TraudASTERISK-27712: [patch] BuildSystem: Detect whether uselocale(.) is available.
Reported by: Alexander TraudASTERISK-27711: [patch] BuildSystem: Avoid re-defining of pthread_* on NetBSD.
Reported by: Alexander TraudASTERISK-27710: [patch] BuildSystem: Install init scripts on openSUSE Tumbleweed.
Reported by: Alexander TraudASTERISK-27709: [patch] BuildSystem: Avoid == for comparison in ./configure.
Reported by: Alexander TraudASTERISK-27681: [patch] BuildSystem: Enable IMAP storage on OpenBSD.
Reported by: Alexander TraudASTERISK-27677: [patch] BuildSystem: Enable system provided libedit on OpenBSD.
Reported by: Alexander TraudASTERISK-27670: [patch] BuildSystem: Remove chan_h323 leftovers.
Reported by: Alexander TraudASTERISK-27595: [patch] BuildSystem: Invoke ldconfig with previous paths.
Reported by: Alexander TraudASTERISK-27631: [patch] BuildSystem: Do not warn when bash is not installed.
Reported by: Alexander TraudASTERISK-27634: Determine if the internal editline and stdtime libraries are still relevant
Reported by: George JosephASTERISK-27619: Build System: Require compiler to provide built-in support for atomic references.
Reported by: Corey FarrellASTERISK-27637: [patch] BuildSystem: Enable autotools in FreeBSD.
Reported by: Alexander TraudASTERISK-16951: [patch] configure.ac in 1.4.37 broken with autoconf 2.60
Reported by: Stéphan KochenASTERISK-27602: [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
Reported by: Alexander TraudASTERISK-27600: [patch] BuildSystem: Allow make clean all again.
Reported by: Alexander TraudASTERISK-27596: [patch] BuildSystem: Use the detected name for MD5 everywhere.
Reported by: Alexander TraudASTERISK-27594: [patch] BuildSystem: Invoke install not in GNU but POSIX style.
Reported by: Alexander TraudASTERISK-27593: [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
Reported by: Alexander TraudASTERISK-27592: [patch] BuildSystem: Detect external library Lua in version 5.3.
Reported by: Alexander TraudASTERISK-27589: [patch] BuildSystem: Avoid $EUID and use id -u instead.
Reported by: Alexander TraudASTERISK-27585: [patch] BuildSystem: Resolve resolv.h not via Generic but Particular Header-Check.
Reported by: Alexander TraudASTERISK-27575: menuselect : remove obsolete TRACE_FRAMES compiler flag
Reported by: Jean Aunis - PrescomASTERISK-27560: [patch] clang 5 does not know -Wno-format-truncation
Reported by: Alexander TraudASTERISK-25329: Asterisk configure fails on 'cannot find ptlib-config', despite ptlib-config existing
Reported by: Rusty NewtonASTERISK-26046: [patch] Avoid obsolete warnings on autoconf.
Reported by: Alexander TraudASTERISK-27332: Asterisk fails to configure on MacOS Sierra
Reported by: Ivan LarionovASTERISK-26639: core: Disabling xmldoc support does not work. Also results in abort during Asterisk startup.
Reported by: Mr DiniASTERISK-27189: Make --with-pjproject-bundled the default for Asterisk 15
Reported by: George JosephASTERISK-27156: Asterisk won't compile on Fedora 26 with devmode enabled.
Reported by: Corey FarrellASTERISK-26705: libasteriskssl.so not found when asterisk is installed for the 1st time
Reported by: George JosephASTERISK-26872: Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
Reported by: Matt JordanASTERISK-26812: [patch] Fix download_externals To Allow The Use Of curl Or wget
Reported by: Michael L. YoungASTERISK-26802: [patch] Integrity Check Of PJSIP Download Fails
Reported by: Michael L. YoungASTERISK-26109: Asterisk fails building with OpenSSL 1.1.0
Reported by: Tzafrir CohenASTERISK-26608: Compile and link failures on OpenBSD
Reported by: snuffyASTERISK-26592: Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
Reported by: George JosephASTERISK-26546: mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
Reported by: Tzafrir CohenASTERISK-22480: Embedded pjproject: build.mak contains hardcoded full path to version.mak
Reported by: Matt JordanASTERISK-26356: menuselect: invalid test for GTK2
Reported by: Tzafrir CohenASTERISK-26303: [patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
Reported by: Alexander TraudASTERISK-26038: 'make install' doesn't seem to install OS/X init files
Reported by: Tzafrir CohenASTERISK-25289: Build System does not respect CFLAGS and CXXFLAGS when building menuselect
Reported by: Jeffrey WaltonASTERISK-26157: Build: Fix errors highlighted by GCC 6.x
Reported by: George JosephASTERISK-26091: [patch] ar cru creates warning, instead use ar cr
Reported by: Alexander TraudASTERISK-25730: build: make uninstall after make distclean tries to remove root
Reported by: George JosephASTERISK-25434: Compiler flags not reported in 'core show settings' despite usage during compilation
Reported by: Rusty NewtonASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG enabled
Reported by: yaron nahumASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1
Reported by: Stefan EngströmASTERISK-25074: Regression: Recent clang-related change broke cross compiling of Asterisk
Reported by: Sebastian KemperASTERISK-25027: Build System: Many ARI modules are missing dependencies.
Reported by: Corey FarrellASTERISK-25028: Build System: Unneeded defines in asterisk/buildopts.h
Reported by: Corey FarrellASTERISK-25026: Git conversion: Non-C files not switched to ASTERISK_REGISTER_FILE
Reported by: Corey FarrellASTERISK-24954: Git migration: Asterisk version numbers are incompatible with the Test Suite
Reported by: Matt JordanASTERISK-24932: Asterisk 13.x does not build with GCC 5.0
Reported by: Jeffrey C. OllieASTERISK-24880: [patch]Compilation under OpenBSD
Reported by: snuffyASTERISK-20399: Compilation on some systems requires the -fnested-functions flag
Reported by: David M. LeeASTERISK-20850: [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
Reported by: Diederik de GrootASTERISK-18105: most of asterisk modules are unbuildable in cygwin environment
Reported by: feyfreASTERISK-24544: Compile fails on OSX Yosemite because of incorrect detection of htonll and ntohll
Reported by: George JosephASTERISK-23991: [patch]asterisk.pc file contains a small error in the CFlags returned
Reported by: Diederik de GrootASTERISK-24502: Build fails when dev-mode, dont optimize and coverage are enabled
Reported by: Corey FarrellASTERISK-13797: [patch] relax badshell tilde test
Reported by: Tzafrir Cohen

Category: Core/CallCompletionSupplementaryServices

ASTERISK-22732: Deadlock potential in res_fax and CCSS with local channels.
Reported by: Richard MudgettASTERISK-24142: CCSS: crash during shutdown due to device lookup in destroyed container
Reported by: David Brillert

Category: Core/CallerID

ASTERISK-24406: Some caller ID strings are parsed differently since 11.13.0
Reported by: Etienne Lessard

Category: Core/Channels

ASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases
Reported by: Mohit DhimanASTERISK-28089: function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload
Reported by: Emmanuel BUUASTERISK-27625: channels: CHECK_BLOCKING is ineffective
Reported by: Corey FarrellASTERISK-27743: Generic PLC doesn't work if the 2 codecs on a channel are equal
Reported by: George JosephASTERISK-25128: Datastore: Implement automatic module references.
Reported by: Corey FarrellASTERISK-27180: channel: requester leaks joint_cap on success.
Reported by: Corey FarrellASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an error branch.
Reported by: Corey FarrellASTERISK-27074: core_local: local channel data not being properly unref'ed and unlocked
Reported by: Kevin HarwellASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge
Reported by: Torrey SearleASTERISK-27025: channel / meetme: Fix missing parentheses
Reported by: Joshua C. ColpASTERISK-26331: Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
Reported by: CGI.NETASTERISK-26306: channel: Hang-up crashes, chan_pjsip not cleaning up properly
Reported by: Alexander TraudASTERISK-25690: Hanging up when executing connected line sub does not cause hangup
Reported by: Joshua C. ColpASTERISK-24991: Check for ao2_alloc failure in __ast_channel_internal_alloc
Reported by: Corey FarrellASTERISK-24380: core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
Reported by: Matt JordanASTERISK-21038: CLI: "core set debug channel" auto-complete returns "all", but not the names of available channels
Reported by: Richard KennerASTERISK-24828: Fix Frame Leaks
Reported by: Kevin HarwellASTERISK-24542: [patch]Failure showing codecs via 'core show channeltype '
Reported by: snuffy

Category: Core/CodecInterface

ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
Reported by: Richard MudgettASTERISK-25172: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request
Reported by: Matt JordanASTERISK-21777: Asterisk tries to transcode video instead of audio
Reported by: Nick RugglesASTERISK-16779: Cannot disallow unknown format ''
Reported by: Atis LezdinsASTERISK-24796: Codecs and bucket schema's prevent module unload
Reported by: Corey FarrellASTERISK-24604: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core
Reported by: Matt Jordan

Category: Core/Configuration

ASTERISK-28158: Some conditions prevent running of el_end, break the terminal.
Reported by: Corey FarrellASTERISK-27863: config/ast_destroy_realtime_fields: successful DELETE is treated as failed
Reported by: Alexei GradinariASTERISK-27415: asterisk.conf: Setting astctl without setting astrundir is ineffective.
Reported by: Corey FarrellASTERISK-27318: res_pjsip_mwi: uninitialized value from ast_strings_match
Reported by: Corey FarrellASTERISK-25956: Compilation error in conditionally compiled code in config_options.c
Reported by: Chris TrobridgeASTERISK-25868: Sorcery "append to category" should allow filters
Reported by: Nick RepinASTERISK-25612: Configuration parser handles unsigned integers as signed integers
Reported by: Gianluca MerloASTERISK-25725: core: Incorrect XML documentation may result in weird behavior
Reported by: Joshua C. ColpASTERISK-25700: main/config: Clean config maps on shutdown.
Reported by: Corey FarrellASTERISK-25683: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG
Reported by: yaron nahumASTERISK-25042: asterisk.conf options override command-line options.
Reported by: Corey FarrellASTERISK-24231: crash: CLI execution of realtime destroy sippeers id 1 causes crash due to NULL name provided to ast_variable
Reported by: Niklas LarssonASTERISK-23733: 'reload acl' fails if acl.conf is not present on startup
Reported by: Richard KennerASTERISK-23651: Reloading some modules that are loaded already, results in 'No such module' before a successful reload
Reported by: Rusty NewtonASTERISK-20127: [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
Reported by: George JosephASTERISK-24487: configuration: sections should be loadable as template even when not marked
Reported by: Scott Griepentrog

Category: Core/DNS

ASTERISK-27495: DNS: Unexpected rr_type can cause crash
Reported by: Corey FarrellASTERISK-26772: Crash in srv.c on startup with pjsip
Reported by: nappsoftASTERISK-25565: DNS: System resolver only returns 1 record per result
Reported by: George Joseph

Category: Core/Dial

ASTERISK-26959: dial: Allow topology of dialing channel to influence dialed channel
Reported by: Joshua C. Colp

Category: Core/FileFormatInterface

ASTERISK-25998: file: Crash when using nativeformats
Reported by: Joshua C. ColpASTERISK-24492: main/file.c: ast_filestream sometimes causes extra calls to ast_module_unref
Reported by: Corey Farrell

Category: Core/General

ASTERISK-28232: core: RAII using clang use-after-scope issue
Reported by: Diederik de GrootASTERISK-28158: Some conditions prevent running of el_end, break the terminal.
Reported by: Corey FarrellASTERISK-28005: channel.c: ARI ring only once
Reported by: Hajek MichalASTERISK-12382: menuselect compilation failure on Solaris 10 / gcc 3.4.3
Reported by: rleasureASTERISK-9107: menuselect compilation failure on Solaris 10/gcc-4.1.1
Reported by: Bob AtkinsASTERISK-27965: module: Remove old modules, update support levels
Reported by: Joshua C. ColpASTERISK-27876: [patch] tcptls: Allow OpenSSL configured with no-dh.
Reported by: Alexander TraudASTERISK-27874: [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated.
Reported by: Alexander TraudASTERISK-27865: [patch]: tcptls: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-27773: Command line not being parsed correctly with getopt not from glibc
Reported by: Guido FalsiASTERISK-24488: Wrong remote identity and target in dialog package XML in NOTIFY
Reported by: Alejandro PadillaASTERISK-26563: core: macOS devmode build fails: variable 'freeswap' set but not used
Reported by: David M. LeeASTERISK-27620: New module loader aborts startup if a required module declines load.
Reported by: snuffyASTERISK-27534: chan_sip: Assumes iostream is non-NULL when it may not be
Reported by: Lubos DolezelASTERISK-27531: Compiler optimizations can break module load sequence.
Reported by: abelbeckASTERISK-27412: core: Audiohook freeing interpolated frame when it shouldn't.
Reported by: MikhailASTERISK-27415: asterisk.conf: Setting astctl without setting astrundir is ineffective.
Reported by: Corey FarrellASTERISK-27404: DEBUG_FD_LEAKS does not record socketpair, timerfd_create or eventfd.
Reported by: Corey FarrellASTERISK-27394: [patch] tcptls: Print notice when TLS is enabled but not configured.
Reported by: Alexander TraudASTERISK-27378: Modules: Fix issues with CLI completion.
Reported by: Corey FarrellASTERISK-27390: Audit menuselect module dependencies
Reported by: Corey FarrellASTERISK-27317: vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
Reported by: Corey FarrellASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-27222: core: Don't queue up multiple video update frames.
Reported by: Joshua C. ColpASTERISK-26745: Asymmetric codecs when asymmetric_rtp_codec=no
Reported by: Jesse RossASTERISK-27105: [patch]core: when setting 'maxfiles' in asterisk.conf, a message is printed, even in rasterisk -x
Reported by: Tzafrir CohenASTERISK-26789: Audit manipulation of channel flags without locks
Reported by: Joshua C. ColpASTERISK-26606: tcptls: Incorrect OpenSSL function call leads to misleading error report
Reported by: Bob HamASTERISK-26528: [UBSAN] strings.h:signed integer overflow in ast_str_case_hash
Reported by: Badalian VyacheslavASTERISK-26903: Listening TCP/TLS sockets stop when temporarily out of open files
Reported by: Walter DoekesASTERISK-26885: channel: Support dynamic number of file descriptors
Reported by: Joshua C. ColpASTERISK-26839: core: Implement stream topology changing in channels
Reported by: Joshua C. ColpASTERISK-26811: stream: Add streams to "core show channel"
Reported by: Joshua C. ColpASTERISK-26786: Implement ast_stream_topology API
Reported by: George JosephASTERISK-26788: core: Protect flags during ast_waitfor
Reported by: Joshua C. ColpASTERISK-26773: stream: Add basic API
Reported by: Joshua C. ColpASTERISK-26632: core: Possibility of a frame "imbalance" leading to stuck channels.
Reported by: Mark MichelsonASTERISK-25083: Message.c: Message channel becomes saturated with frames leading to spammy log messages
Reported by: Jonathan RoseASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
Reported by: Richard MudgettASTERISK-26509: A few non-critical deprecation warnings when building on Ubuntu 16.10
Reported by: Jonathan HarrisASTERISK-26466: core: Be forgiving on external callerid that may be flawed so we don't drop events
Reported by: Richard MudgettASTERISK-26273: core: Won't compile when LOW_MEMORY is enabled
Reported by: Anthony MessinaASTERISK-26331: Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
Reported by: CGI.NETASTERISK-26267: ast_register_atexit callbacks should be run on failed startup.
Reported by: Corey FarrellASTERISK-26253: sdp_srtp: libsrtp now a required dependency, shouldn't be
Reported by: Ben MerrillsASTERISK-26278: asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
Reported by: Corey FarrellASTERISK-26265: Errors ignored from some parts of system initialization.
Reported by: Corey FarrellASTERISK-25996: Remove "live_dangerously" requirement on DB(read)
Reported by: Andrew NagyASTERISK-26237: Fax is detected on regular calls.
Reported by: Richard MudgettASTERISK-14: asterisk leaves zombie mpg123
Reported by: dcarrASTERISK-26191: threadpool: Leak on duplicate taskprocessor for ast_threadpool_serializer_group
Reported by: Corey FarrellASTERISK-26119: [patch] fix: memory leaks, resource leaks, out of bounds and bugs
Reported by: Alexei GradinariASTERISK-26097: [patch] CLI: show maximum file descriptors
Reported by: Alexander TraudASTERISK-25894: [patch] webrtc video broken due to missing marker bits in RTP streams
Reported by: Jacek KoniecznyASTERISK-25825: Crashes during shutdown when running CLI commands
Reported by: Mark MichelsonASTERISK-25681: devicestate: Engine thread is not shut down
Reported by: Corey FarrellASTERISK-25307: Hangup on channel using FastAGI does not hang up child channels
Reported by: David CunninghamASTERISK-25601: json: Audit reference usage and thread safety
Reported by: Joshua C. ColpASTERISK-25585: [patch]rasterisk never hits most of main(), but it's assumed to
Reported by: Walter DoekesASTERISK-25552: hashtab: Improve NULL tolerance
Reported by: Joshua C. ColpASTERISK-25449: main/sched: Regression introduced by 5c713fdf18f causes erroneous duplicate RTCP messages; other potential scheduling issues in chan_sip/chan_skinny
Reported by: Matt JordanASTERISK-25546: threadpool: Race condition between idle timeout and activation
Reported by: Joshua C. ColpASTERISK-25528: DNS: System resolver issues with TTL parse
Reported by: dtrybaASTERISK-7803: [patch] Update the maximum packetization values in frame.c
Reported by: deaASTERISK-25383: Core dumps on startup and shutdown with MALLOC_DEBUG enabled
Reported by: yaron nahumASTERISK-25418: On-hold channels redirected out of a bridge appear to still be on hold
Reported by: Mark MichelsonASTERISK-25355: sched: ast_sched_del may return prematurely due to spurious wakeup
Reported by: Joshua C. ColpASTERISK-25255: Missing AMI VarSet events when setting to an empty string.
Reported by: Richard MudgettASTERISK-25201: Crash in PJSIP distributor on already free'd threadpool
Reported by: Matt JordanASTERISK-25146: DNS: Create system level resolver
Reported by: Joshua C. ColpASTERISK-25222: Crash in recurring cancel callback called from ast_dns_resolve_cancel on junk pointer
Reported by: Matt JordanASTERISK-25212: [patch]Segfault when using DEBUG_FD_LEAKS
Reported by: Walter DoekesASTERISK-22559: gcc 4.6 and higher supports weakref attribute but asterisk doesn't detect it.
Reported by: ibercomASTERISK-24944: main/audiohook.c change prevents G722 call recording
Reported by: Ronald RaikesASTERISK-24896: [patch] Using force black background leads to colours not being reset
Reported by: dantASTERISK-24997: Astobj2: Some callers of __adjust_lock do not pre-check the object
Reported by: Corey FarrellASTERISK-24994: dns: Query set unit tests are failing due to race condition
Reported by: Joshua C. ColpASTERISK-24155: [patch]Non-portable and non-reliable recursion detection in ast_malloc
Reported by: Timo TeräsASTERISK-24881: ast_register_atexit should only be used when absolutely needed
Reported by: Corey FarrellASTERISK-24879: [patch]Compilation fails due to 64bit time under OpenBSD
Reported by: snuffyASTERISK-24739: [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules
Reported by: Ed HynanASTERISK-24796: Codecs and bucket schema's prevent module unload
Reported by: Corey FarrellASTERISK-24814: asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64 bit integers
Reported by: Corey FarrellASTERISK-24740: [patch]Segmentation fault on aoc-e event
Reported by: Panos GkikakisASTERISK-24752: Crash in bridge_manager_service_req when bridge is destroyed by ARI during shutdown
Reported by: Richard MudgettASTERISK-24479: Enable REF_DEBUG for module references
Reported by: Corey FarrellASTERISK-24736: Memory Leak Fixes
Reported by: Mark MichelsonASTERISK-24619: [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly casts char to unsigned int
Reported by: Walter DoekesASTERISK-24614: Deadlock when DEBUG_THREADS compiler flag enabled
Reported by: Richard MudgettASTERISK-23651: Reloading some modules that are loaded already, results in 'No such module' before a successful reload
Reported by: Rusty NewtonASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks
Reported by: Corey FarrellASTERISK-24348: Built-in editline tab complete segfault with MALLOC_DEBUG
Reported by: Walter Doekes

Category: Core/HTTP

ASTERISK-26794: http: Crash on Reload Only in ast_tcptls_server_start
Reported by: Joshua ElsonASTERISK-26126: [patch] leverage 'bindaddr' for TLS in http.conf
Reported by: Alexander TraudASTERISK-24724: 'httpstatus' Web Page Produces Incomplete HTML
Reported by: Ashley Sanders

Category: Core/Jitterbuffer

ASTERISK-27194: jitterbuffer: Does not handle case where translator returns null frame.
Reported by: Joshua ElsonASTERISK-22409: Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade
Reported by: Corey Farrell

Category: Core/Logging

ASTERISK-23462: Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't
Reported by: Rusty NewtonASTERISK-27340: backtrace.c: Crash due to double-free.
Reported by: Corey FarrellASTERISK-26410: core: Asterisk 14 doesn't show the header in the console or verbose when starting
Reported by: Dan JenkinsASTERISK-26078: core: Memory leak in logging
Reported by: Etienne LessardASTERISK-25538: [patch]Missing PID in syslog logger messages
Reported by: Javier AcostaASTERISK-25407: Asterisk fails to log to multiple syslog destinations
Reported by: Elazar BroadASTERISK-25510: [patch]Log to syslog failing
Reported by: Michael NewtonASTERISK-24833: [patch] audit of startup order reveals logger concerns
Reported by: Corey FarrellASTERISK-25305: Dynamic logger channels can be added multiple times
Reported by: Mark MichelsonASTERISK-25112: Logger: Configuration settings are not reset to default during reload.
Reported by: Corey FarrellASTERISK-24817: init_logger_chain: unreachable code block
Reported by: Corey FarrellASTERISK-24223: Gibberish Call-ID on Local channel on origination
Reported by: Mark Michelson

Category: Core/ManagerInterface

ASTERISK-28350: manager: Stasis backed up due to locking
Reported by: Joshua C. ColpASTERISK-28084: app_queue: QueueMemberStatus Event flooding AMI
Reported by: AndrejASTERISK-28033: AMI event "NewExten" is set to the wrong class
Reported by: lvlASTERISK-27943: AMI: Action SendText needs to use the correct thread.
Reported by: Richard MudgettASTERISK-27852: cli: "manager show settings" mislabels HTTP timeout as being minutes.
Reported by: Corey FarrellASTERISK-27841: digest over for manager (ami) over http fails on too long uris
Reported by: Jaco KroonASTERISK-27659: Output from rawman truncated if output is long enough
Reported by: Bojan NemčićASTERISK-27200: manager: hook event is not being raised
Reported by: Kevin HarwellASTERISK-27073: manager: AMI "queues" action outputs freeform text that doesn't follow the AMI spec
Reported by: BrianASTERISK-26629: tests/manager: 4 test failures as a result of iostream change
Reported by: Joshua C. ColpASTERISK-26556: manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
Reported by: Michelle DupuisASTERISK-26537: AMI: NewConnectedLine event is not documented
Reported by: Etienne LessardASTERISK-26397: manager: PresenceState action crashes Asterisk 14
Reported by: Andrew NagyASTERISK-26246: Security: Privilege escalation by AMI adding dialplan extensions.
Reported by: Richard MudgettASTERISK-25680: manager: manager_channelvars is not cleaned at shutdown
Reported by: Corey FarrellASTERISK-25624: AMI Event OriginateResponse bug
Reported by: sungtae kimASTERISK-25391: AMI GetConfigJSON returns invalid JSON
Reported by: Bojan NemčićASTERISK-24934: [patch]Asterisk manager output does not escape control characters
Reported by: warren smithASTERISK-24900: Manager event ParkedCallSwap is not documented
Reported by: Rusty NewtonASTERISK-20524: AMI improperly handles lines of exactly 1025 characters
Reported by: David M. LeeASTERISK-22670: Asterisk crashes when processing ISDN AoC Events
Reported by: klaus3000ASTERISK-24721: manager: ModuleLoad action incorrectly reports 'module not found' during a Reload operation
Reported by: Matt JordanASTERISK-24049: Asterisk Manager Interface: A number of list type responses aren't using astman_send_listack
Reported by: Jonathan RoseASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
Reported by: Niklas LarssonASTERISK-24505: manager: http connections leak references
Reported by: Corey FarrellASTERISK-22409: Local channels in a ConfBridge w/ jitterbuffer=yes leak ast_frame's after masquerade
Reported by: Corey FarrellASTERISK-24453: manager: acl_change_sub leaks
Reported by: Corey FarrellASTERISK-24430: missing letter "p" in word response in OriginateResponse event documentation
Reported by: Dafi NiASTERISK-24354: AMI sendMessage closes AMI connection on error
Reported by: Peter KatzmannASTERISK-24378: Release AMI connections on shutdown
Reported by: Corey FarrellASTERISK-24262: AMI CoreShowChannel missing several output fields and event documentation
Reported by: Mitch ClabornASTERISK-24331: Unexpected Errors in Asterisk Manager Interface Output
Reported by: xrobauASTERISK-24138: dial: Call forwarding information presented through AMI/ARI is wrong
Reported by: Matt Jordan

Category: Core/ManagerInterface/NewFeature

ASTERISK-25624: AMI Event OriginateResponse bug
Reported by: sungtae kimASTERISK-25189: AMI: Add Linkedid header to standard channel snapshot information.
Reported by: Richard Mudgett

Category: Core/Netsock

ASTERISK-24469: Security Vulnerability: Mixed IPv4/IPv6 ACLs allow blocked addresses through
Reported by: Matt Jordan

Category: Core/PBX

ASTERISK-28300: AST_PBX_MAX_STACK is too low for some applications
Reported by: George JosephASTERISK-28140: repeated segmentation faults
Reported by: Eyal HassonASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
Reported by: Frederic LE FOLLASTERISK-26115: pbx: AMI Originate ignore "failed" extension on call failure
Reported by: Nasir IqbalASTERISK-26226: pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
Reported by: Etienne LessardASTERISK-26233: pbx: Failure to remove inconsistent extension names
Reported by: Corey FarrellASTERISK-26196: pbx: Time based includes can leak timezone string
Reported by: Corey FarrellASTERISK-25881: pbx: Add support for autohints
Reported by: Joshua C. ColpASTERISK-25394: pbx: Incorrect device and presence state when changing hint details
Reported by: Joshua C. ColpASTERISK-25367: pbx: Long pattern match hints may cause "core show hints" to crash
Reported by: Joshua C. ColpASTERISK-25094: PBX core: Investigate thread safety issues
Reported by: Corey FarrellASTERISK-24442: Outgoing call files don't work properly when set in the future
Reported by: tootaiASTERISK-24774: Segfault in ast_context_destroy with extensions.ael and extensions.conf
Reported by: Corey FarrellASTERISK-24914: Division by zero in file.c when playback of voicemail with video as h264
Reported by: Marcello CeschiaASTERISK-24683: Crash in PBX ast_hashtab_lookup_internal during core restart now
Reported by: Peter KatzmannASTERISK-24805: [patch] - ASAN: Race condition (heap-use-after-free) on asterisk closing
Reported by: Badalian VyacheslavASTERISK-24641: Deadlock in Trunk
Reported by: Malcolm DavenportASTERISK-24444: PBX: Crash when generating extension for pattern matching hint
Reported by: Leandro DardiniASTERISK-24249: SIP debugs do not stop
Reported by: Avinash Mohod

Category: Core/Portability

ASTERISK-15331: make menuselect fails due to undefined symbols (initscr32, w32addch) in menuselect_curses.o
Reported by: Majdi BsoulASTERISK-14935: [regression] menuselect compilation failure on Solaris 10
Reported by: Samuel OwensASTERISK-27933: [patch] uuid: Enable UUID in Solaris 11.
Reported by: Alexander TraudASTERISK-27431: Asterisk fails to build when openssl headers are not installed.
Reported by: Corey FarrellASTERISK-24515: Unconditional use of fopencookie() / funopen() is non-portable
Reported by: Timo TeräsASTERISK-24155: [patch]Non-portable and non-reliable recursion detection in ast_malloc
Reported by: Timo Teräs

Category: Core/RTP

ASTERISK-27854: rtp: Crash in off-nominal case where RTP instance can't be set up
Reported by: Lei FuASTERISK-27967: srtp: rejecting short sdes lifetimes incompatible with obihai ATAs
Reported by: Nick FrenchASTERISK-27831: res_rtp_asterisk: Add support for abs-send-time RTP extension
Reported by: Joshua C. ColpASTERISK-27850: [patch] rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again.
Reported by: Alexander TraudASTERISK-27689: [patch] rtp_engine: Load format name / mime type in uppercase again.
Reported by: Alexander TraudASTERISK-27225: Crash when freeing dtls_cfg->cafile
Reported by: Richard KennerASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code()
Reported by: Ross BeerASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
Reported by: Frankie ChinASTERISK-26515: rtp_engine: Allocate RTP payloads on a per-session basis
Reported by: Joshua C. ColpASTERISK-24274: [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
Reported by: Frankie ChinASTERISK-26311: [patch] rtp_engine: Allow more than 32 dynamic payload types.
Reported by: Alexander TraudASTERISK-26365: rtp: Offer with multiple payloads for same codec is incorrectly handled
Reported by: Joshua C. ColpASTERISK-26367: rtp: Timestamps broken when video frame is across multiple RTP packets
Reported by: Joshua C. ColpASTERISK-25296: RTP performance issue with several channel drivers.
Reported by: Richard MudgettASTERISK-25219: [patch]Source and destination overlap in memcpy in rtp_engine.c
Reported by: Walter DoekesASTERISK-25022: Memory leak setting up DTLS/SRTP calls
Reported by: Steve DaviesASTERISK-24489: Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
Reported by: Gregory Malsack

Category: Core/SQLite3

ASTERISK-25996: Remove "live_dangerously" requirement on DB(read)
Reported by: Andrew Nagy

Category: Core/Sorcery

ASTERISK-27972: res_sorcery_config: Allow object name based matching
Reported by: Joshua C. ColpASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c
Reported by: Ryan SmithASTERISK-26172: res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows.
Reported by: Alexei GradinariASTERISK-26014: res_sorcery_astdb: Make tolerant of unknown fields
Reported by: Joshua C. ColpASTERISK-25826: PJSIP / Sorcery slow load from realtime
Reported by: Ross BeerASTERISK-25811: Unable to delete object from sorcery cache
Reported by: Ross BeerASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic ColledgeASTERISK-25625: res_sorcery_memory_cache: Add full backend caching
Reported by: Joshua C. ColpASTERISK-25165: Testsuite - Sorcery memory cache leaks
Reported by: Corey FarrellASTERISK-25141: pjsip_options: Contact reference leak
Reported by: Corey FarrellASTERISK-24996: chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf
Reported by: Ashley SandersASTERISK-24612: res_pjsip: No information if a required sorcery wizard is not loaded
Reported by: Joshua C. ColpASTERISK-24312: SIGABRT when improperly configured realtime pjsip
Reported by: Dafi Ni

Category: Core/Stasis

ASTERISK-28335: stasis: Make topic and maybe subscription names unique and more useful
Reported by: Joshua C. ColpASTERISK-28252: HangupHandler manager events are never thrown
Reported by: Gerald SchnabelASTERISK-28244: stasis: Filter messages at publishing to AMI/ARI
Reported by: Joshua C. ColpASTERISK-28197: stasis: ast_endpoint struct holds the channel_ids of channels past destruction in certain cases
Reported by: Mohit DhimanASTERISK-28212: stasis: Statistics broke ABI under developer mode
Reported by: Joshua C. ColpASTERISK-28117: stasis: Add statistics for usage when in developer mode
Reported by: Joshua C. ColpASTERISK-28186: stasis: Filter messages at publishing based on to_* presence
Reported by: Joshua C. ColpASTERISK-28103: stasis: Filter messages at publishing to reduce work done
Reported by: Joshua C. ColpASTERISK-28084: app_queue: QueueMemberStatus Event flooding AMI
Reported by: AndrejASTERISK-27591: Frack errors in stasis.c and memory leakage
Reported by: Siruja MaharjanASTERISK-25548: stasis: Improve message type "Use of before init/after destruction" error
Reported by: Joshua C. ColpASTERISK-25237: stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
Reported by: Smirnov AlekseyASTERISK-26468: ari: Bridge events stop working after this sequence of ARI calls
Reported by: Daniele PallastrelliASTERISK-25137: endpoint stasis messages are delivered twice
Reported by: Vitezslav NovyASTERISK-25121: Stasis: Fix unsafe use of stasis_unsubscribe in modules.
Reported by: Corey FarrellASTERISK-24682: app_dial: Multiple DialEnd events emitted when MACRO_RESULT or GOSUB_RESULT are an unexpected value
Reported by: Matt Jordan

Category: Core/Streams

ASTERISK-27488: core: If frame with unnegotiated format is read crash will occur
Reported by: Sébastien DuthilASTERISK-27379: stream: Allow streams on a topology to be put into groups
Reported by: Joshua C. Colp

Category: Core/UDPTL

ASTERISK-26034: T.38 passthrough problem behind firewall due to early nosignal packet
Reported by: George JosephASTERISK-25603: [patch]udptl: Uninitialized lengths and bufs in udptl_rx_packet cause ast_frdup crash
Reported by: Walter DoekesASTERISK-25742: Secondary IFP Packets can result in accessing uninitialized pointers and a crash
Reported by: Torrey Searle

Category: Documentation

ASTERISK-20986: QUEUE_MEMBER 's description is inaccurate
Reported by: Olivier KriefASTERISK-24173: File menuselect/menuselect_gtk.c has no license header
Reported by: Jeremy LainéASTERISK-28150: Formatting error in documentation
Reported by: Scott GriepentrogASTERISK-25261: Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User'
Reported by: Francois BlackburnASTERISK-26688: Documentation: voicemail.conf.sample shows 512 limit for emailbody field, however this is only true if compiled with LOW_MEMORY option
Reported by: Fran VicenteASTERISK-24386: Asterisk "doc/lang/language-criteria.txt" needs update or removal.
Reported by: Rusty NewtonASTERISK-24198: Typo's
Reported by: Walter DoekesASTERISK-25649: Transfer application does not work with Local channels - documentation misleading
Reported by: Ivan UllmannASTERISK-27430: README refers to security documents that do not exist.
Reported by: Corey FarrellASTERISK-27377: Typo in CHANNEL(dtmf_features) usage documentation
Reported by: Igor GoncharovskyASTERISK-25523: res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
Reported by: JesperASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP argument
Reported by: Rusty NewtonASTERISK-26086: res_musiconhold: format option is not documented adequately
Reported by: Jens BürgerASTERISK-26484: res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
Reported by: Vinod DharashiveASTERISK-26717: Document the fact that Asterisk HEP support only works with the PJSIP channel driver
Reported by: Olivier KriefASTERISK-25237: stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
Reported by: Smirnov AlekseyASTERISK-24562: app_voicemail: Cannot set fromstring on a per-mailbox basis
Reported by: Mark ScholtenASTERISK-26782: res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
Reported by: Peter SokolovASTERISK-26704: res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
Reported by: Anthony MessinaASTERISK-26514: Super Awesome Company: Don't specify transport in pjsip.conf
Reported by: Rusty NewtonASTERISK-25472: Swagger scripts are not replacing format variable in file brief
Reported by: Corey FarrellASTERISK-26212: [patch] Makefile: Retain XML Declaration and DTD in docs.
Reported by: Alexander TraudASTERISK-25927: Removed option "registertrying" is still documented in sip.conf.sample
Reported by: Etienne LessardASTERISK-24097: Documentation - CHANNEL function help text missing 'linkedid' argument
Reported by: Steven WheelerASTERISK-25373: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
Reported by: Walter DoekesASTERISK-25527: Quirky xmldoc description wrapping
Reported by: Walter DoekesASTERISK-24867: Docs for 'e' option in ResetCDR say to use CDR_PROP instead, CDR_PROP docs are unclear
Reported by: Rusty NewtonASTERISK-24853: Documentation claims chan_sip outbound registrations support WS or WSS as valid transports (not true)
Reported by: PSDKASTERISK-24085: Documentation - We should remove or further document the 'contact' section in pjsip.conf
Reported by: Rusty NewtonASTERISK-24430: missing letter "p" in word response in OriginateResponse event documentation
Reported by: Dafi NiASTERISK-24419: Incorrect syntax for setting language in configs/extensions.conf.sample
Reported by: Ben KlangASTERISK-24122: Documentaton for res_pjsip option use_avpf needs to be fixed
Reported by: James Van VleetASTERISK-24262: AMI CoreShowChannel missing several output fields and event documentation
Reported by: Mitch ClabornASTERISK-23768: [patch] Asterisk man page contains a (new) unquoted minus sign
Reported by: Jeremy Lainé

Category: Features

ASTERISK-26781: bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
Reported by: Sean BrightASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John KinistonASTERISK-25003: Asterisk crashes on attended transfer (using feature)
Reported by: Artem VolodinASTERISK-23841: DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.
Reported by: Richard Mudgett

Category: Features/Parking

ASTERISK-26029: parking: ast_parking_park_call should return parking_space instead of parking_exten
Reported by: Diederik de Groot

Category: Formats/General

ASTERISK-27549: [patch] translate: Avoid absolute value on unsigned substraction.
Reported by: Alexander TraudASTERISK-26426: format_ogg_opus: remove from source
Reported by: Kevin HarwellASTERISK-25664: ast_format_cap_append_by_type leaks a reference
Reported by: Corey FarrellASTERISK-25584: [patch] format-attribute module: VP8 missing
Reported by: Alexander TraudASTERISK-25545: [patch] translation module gets cached not joint format
Reported by: Alexander TraudASTERISK-25535: [patch] format creation on module load instead of cache
Reported by: Alexander TraudASTERISK-25537: [patch] format-attribute module: RFC or internal defaults?
Reported by: Alexander TraudASTERISK-25533: [patch] buffer for ast_format_cap_get_names only 64 bytes
Reported by: Alexander TraudASTERISK-25054: Formats interface's cannot be unregistered, needs to hold modules until shutdown.
Reported by: Corey Farrell

Category: Formats/format_h264

ASTERISK-25573: [patch] H.264 format attribute module: resets whole SDP
Reported by: Alexander Traud

Category: Formats/format_ogg_vorbis

ASTERISK-12841: [patch] Make format_ogg_vorbis work on OpenBSD
Reported by: Michiel van BaakASTERISK-26169: format_ogg_vorbis: Memory leak using OGG in MixMonitor
Reported by: Ivan Myalkin

Category: Formats/format_pcm

ASTERISK-20984: Audible clicks when playing sox encoded au file with STREAM FILE AGI command
Reported by: Roman S.

Category: Formats/format_wav

ASTERISK-26613: format_wav: wav16 format read file only by 320 - half of frame
Reported by: Vitaly K

Category: Functions/General

ASTERISK-23133: Documentation fix - MASTER_CHANNEL Unexpected Behaviour
Reported by: Shane MitchellASTERISK-17608: func_aes.so cannot be loaded if res_crypto / openssl not compiled
Reported by: Warren Selby

Category: Functions/func_aes

ASTERISK-27908: [patch] crypto.h: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-25857: func_aes: incorrect use of strlen() leads to data corruption
Reported by: Gianluca Merlo

Category: Functions/func_callerid

ASTERISK-25373: add documentation for CALLERID(pres) and also the CONNECTEDLINE and REDIRECTING variants
Reported by: Walter Doekes

Category: Functions/func_cdr

ASTERISK-27460: CDR: Deadlock using AMI Originate with Variable CDR(amaflags)=...
Reported by: Richard MudgettASTERISK-26173: func_cdr: CDR function does not permit empty values to be assigned
Reported by: gkloepferASTERISK-25179: CDR(billsec,f) and CDR(duration,f) report incorrect values
Reported by: Gianluca MerloASTERISK-24455: func_cdr: CDR_PROP leaks payload
Reported by: Corey Farrell

Category: Functions/func_channel

ASTERISK-24097: Documentation - CHANNEL function help text missing 'linkedid' argument
Reported by: Steven Wheeler

Category: Functions/func_curl

ASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code.
Reported by: Corey FarrellASTERISK-25669: [patch]CURL incorrect trim for non ASCII characters
Reported by: JesperASTERISK-18708: func_curl hangs channel under load
Reported by: Dave CabotASTERISK-24676: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
Reported by: Matt JordanASTERISK-24672: [PATCH] Memory leak in func_curl CURLOPT
Reported by: Kristian Høgh

Category: Functions/func_db

ASTERISK-24534: [patch]Register DB() as escalating to prevent users from writing to astdb
Reported by: Gareth Palmer

Category: Functions/func_devstate

ASTERISK-26643: Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
Reported by: Roman Bedros

Category: Functions/func_dialplan

ASTERISK-21765: [patch] - FILE function's length argument counts from beginning of file rather than the offset
Reported by: John Zhong

Category: Functions/func_iconv

ASTERISK-25272: [patch]The ICONV dialplan function sometimes returns garbage
Reported by: Etienne Lessard

Category: Functions/func_odbc

ASTERISK-27888: SQL fetch error on query which return 0 columns
Reported by: Alexei GradinariASTERISK-25984: res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
Reported by: József DudásASTERISK-26177: func_odbc: Database handle is kept when it should be released
Reported by: Leandro DardiniASTERISK-25938: res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
Reported by: Edwin VandammeASTERISK-25963: func_odbc requires reconnect checks for stale connections
Reported by: Ross BeerASTERISK-22708: res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
Reported by: JoshE

Category: Functions/func_periodic_hook

ASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-25085: [patch]Potential crash after unload of func_periodic_hook or test_message
Reported by: Corey Farrell

Category: Functions/func_shell

ASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey Farrell

Category: Functions/func_speex

ASTERISK-26926: func_speex: Crash caused by frame with no datalen
Reported by: Richard Kenner

Category: Functions/func_strings

ASTERISK-28159: SIGABRT caused by stack corruption in hashkeys_read when no matching keys present
Reported by: Michael WaltonASTERISK-25669: [patch]CURL incorrect trim for non ASCII characters
Reported by: Jesper

Category: Functions/func_talkdetect

ASTERISK-24988: func_talkdetect: Test is bouncing sporadically
Reported by: Joshua C. ColpASTERISK-24482: func_talkdetect: Fix stasis message leak in audiohook callback
Reported by: Corey Farrell

Category: General

ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
Reported by: Ted GASTERISK-28523: Asterisk 16.5.0 Memory leak
Reported by: Cyril RamièreASTERISK-28332: Variable ALTCONF ignored when service is used in Debian
Reported by: Cirillo FerreiraASTERISK-26366: rtp: RTCP messages with REMB trigger fast picture update
Reported by: Joshua C. ColpASTERISK-27642: [patch] backtrace: Avoid -Wlogical-not-parentheses.
Reported by: Alexander TraudASTERISK-27630: [patch] editline: Avoid shifting a negative signed value.
Reported by: Alexander TraudASTERISK-27559: [patch] editline: Avoid comparison between pointer and zero character constant.
Reported by: Alexander TraudASTERISK-20346: Modules need to ensure that any functions, apps, AMI actions, etc. they register are unregistered if the module declines loading
Reported by: Mark MichelsonASTERISK-27382: crash after an invalid rtcp packet from GT48 FXS gateway
Reported by: Tzafrir CohenASTERISK-27467: pjsip_options: qualify_frequency sometimes not applied on reload
Reported by: John BigelowASTERISK-24662: [patch] column and row headers for Signed Linear format variants in output of 'core show translation' are ambiguous
Reported by: Rusty NewtonASTERISK-27442: pjsip: 183 without To tag does not negotiate media
Reported by: Kevin HarwellASTERISK-27337: chan_sip: Security vulnerability with client code header (revisited)
Reported by: Richard MudgettASTERISK-27319: (Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long variable in 64bit machines
Reported by: Kim youngsungASTERISK-27305: res_ari: Memory leaks in ARI when using Content-Type: application/json
Reported by: David HajekASTERISK-27295: Contact is improperly translated after d178f497
Reported by: Sean BrightASTERISK-27260: [pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
Reported by: Daniel HecklASTERISK-27177: ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
Reported by: Tzafrir CohenASTERISK-27152: Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
Reported by: Ross BeerASTERISK-27212: bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
Reported by: Richard MudgettASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
Reported by: Joshua C. ColpASTERISK-27060: Comment typo format_g729.c
Reported by: Matthew FredricksonASTERISK-26983: Crash in Manager Reload when TLS Config Changes
Reported by: Joshua ElsonASTERISK-26860: Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
Reported by: Evers LabASTERISK-26949: sdp: Implement T.38
Reported by: Joshua C. ColpASTERISK-26900: sdp: Add support for connection address management and topology updating
Reported by: Joshua C. ColpASTERISK-26668: core: Malformed pattern matching extension (various factors) results in crash
Reported by: xrobauASTERISK-26816: Implement ast_read_stream in channels
Reported by: Joshua C. ColpASTERISK-26825: pjsip.conf.sample: user_agent: still refers to branch 12
Reported by: Tzafrir CohenASTERISK-26793: Implement ast_write_stream in channels
Reported by: George JosephASTERISK-26790: Implement stream topology (non-change request) API usage in channels
Reported by: George JosephASTERISK-26765: res_resolver_unbound: FRACK! Excessive ref count trap tripped.
Reported by: Richard MudgettASTERISK-26754: build_tools: make_build_h does not handle \ in user name
Reported by: Kirill KatsnelsonASTERISK-26575: testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
Reported by: Joshua C. ColpASTERISK-26546: mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
Reported by: Tzafrir CohenASTERISK-25070: Fix FTBFS on Hurd
Reported by: Gabriele GiaconeASTERISK-26387: Asterisk segfaults shortly after starting even with no active calls.
Reported by: Harley PetersASTERISK-26513: tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
Reported by: Joshua C. ColpASTERISK-26480: [patch] CLI: core set debug: Auto-completes File not Module
Reported by: Alexander TraudASTERISK-26421: Segmentation Fault with ARI originate into mixing bridge with 43 clients
Reported by: Andrew NagyASTERISK-26268: alembic: 'auth_username' not in PJSIP 'identify_by' enum
Reported by: Joshua C. ColpASTERISK-26283: res_resolver_unbound: fails configure on older Ubuntu and CentOS
Reported by: George JosephASTERISK-26227: sqlalchemy error due to long identifier name
Reported by: Mark MichelsonASTERISK-26180: PJSIP: provide valid tcp nodelay option for reuse
Reported by: Scott GriepentrogASTERISK-26132: PJSIP: provide transport type with received messages
Reported by: Scott GriepentrogASTERISK-25777: data race in threadpool
Reported by: Badalian VyacheslavASTERISK-25978: res_pjsip_authenticator_digest: Should not use source port in nonce verification
Reported by: Mark MichelsonASTERISK-25948: ast_pthread_mutex_lock calling ast_reentrancy_lock with lt=0x0
Reported by: Diederik de GrootASTERISK-25714: ASAN:heap-buffer-overflow in logger.c
Reported by: Badalian VyacheslavASTERISK-24801: ASAN: ast_el_read_char stack-buffer-overflow
Reported by: Badalian VyacheslavASTERISK-25614: DTLS negotiation delays
Reported by: Dade BrandonASTERISK-25619: res_chan_stats not sending the correct information to StatsD
Reported by: Tyler CambronASTERISK-25461: Nested dialplan #includes don't work as expected.
Reported by: Richard MudgettASTERISK-25435: Asterisk periodically hangs. UDP Recv-Q greatly exceeds zero.
Reported by: Dmitriy SerovASTERISK-25390: default_from_user can crash with certain configuration backends
Reported by: Mark MichelsonASTERISK-25375: Bad ao2 pointer on snapshot cleanup after creation
Reported by: Scott GriepentrogASTERISK-25365: Persistent subscriptions have extra Content-Length/corrupted messages
Reported by: Mark MichelsonASTERISK-25342: res_pjsip: Repeated usage of pj_gethostip may block
Reported by: Joshua C. ColpASTERISK-25331: install_prereq is not installing sqlite 3 library on CentOS
Reported by: Scott GriepentrogASTERISK-25242: PJSIP: No audio when Asterisk inside NAT and endpoints outside NAT - implement functionality similar to chan_sip 'rtpkeepalive'?
Reported by: Mark MichelsonASTERISK-25162: func_pjsip_aor: Leak of contact in iterator
Reported by: Corey FarrellASTERISK-25148: res_pjsip NULL channel audit
Reported by: Mark MichelsonASTERISK-25004: Crash in authenticated reinvite after originated T.38 FAX
Reported by: Mark MichelsonASTERISK-24975: Enabling 'DEBUG_THREADLOCALS' Causes the Build to Fail
Reported by: Ashley SandersASTERISK-23666: CLONE - nested functions aren't portable
Reported by: Diederik de GrootASTERISK-24830: res_rtp_asterisk.c checks USE_PJPROJECT not HAVE_PJPROJECT
Reported by: Stefan EngströmASTERISK-24751: Integer values in json payload to ARI cause asterisk to crash
Reported by: jeffrey putnamASTERISK-24711: DTLS handshake broken with latest OpenSSL versions
Reported by: Jared BielASTERISK-24728: tcptls: Bad file descriptor error when reloading chan_sip
Reported by: Kevin HarwellASTERISK-24693: Investigate and fix memory leaks in Asterisk
Reported by: Kevin HarwellASTERISK-24624: Transfer to invalid extension results in hung channel.
Reported by: Zane ConkleASTERISK-24663: [patch] Unnamed semaphore autoconf check fails on cross compilation
Reported by: abelbeckASTERISK-24655: res_pjsip_outbound_publish: Hang on shutdown while attempting to publish
Reported by: Kevin HarwellASTERISK-24665: Configure check required for pjsip_get_dest_info()
Reported by: Mark MichelsonASTERISK-22455: Asterisk 12 on Ubuntu Lucid deadlocks with DEBUG_THREADS+OPTIONAL_API enabled
Reported by: David M. LeeASTERISK-24563: Direct Media calls within private network sometimes get one way audio
Reported by: Kevin HarwellASTERISK-24504: chan_console: Fix reference leaks to pvt
Reported by: Corey FarrellASTERISK-24465: audiohooks list leaks reference to formats
Reported by: Corey FarrellASTERISK-24321: SIP deadlock when running automated queues tests
Reported by: Steve PittsASTERISK-24224: When using Bridge() dialplan application, surrogate channel appears in list and call count is inflated.
Reported by: Mark MichelsonASTERISK-20567: bashism in autosupport
Reported by: Tzafrir CohenASTERISK-24328: Use of MixMonitor 'm' option results in 0 duration vm description file
Reported by: Scott GriepentrogASTERISK-24245: gcc 4.1.2 complains of files that do not end with newlines
Reported by: Shaun RuffellASTERISK-24246: Quiet warning about type qualifiers ignored on function return type
Reported by: Shaun RuffellASTERISK-24032: Gentoo compilation emits warning: "_FORTIFY_SOURCE" redefined
Reported by: Kilburn

Category: PBX/pbx_config

ASTERISK-28146: pbx_config: Only the first [globals] section is processed.
Reported by: Corey FarrellASTERISK-25881: pbx: Add support for autohints
Reported by: Joshua C. ColpASTERISK-25061: pbx_config: Register manager actions with module version of macro.
Reported by: Corey Farrell

Category: PBX/pbx_dundi

ASTERISK-26987: pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers
Reported by: Kirsty TyermanASTERISK-27908: [patch] crypto.h: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-18731: [patch] DUNDi weight parameter not processed correctly
Reported by: Peter RaczASTERISK-25677: pbx_dundi: leaks during failed load.
Reported by: Corey Farrell

Category: PBX/pbx_lua

ASTERISK-27553: [patch] res_curl: Avoid error message on unload.
Reported by: Alexander Traud

Category: PBX/pbx_realtime

ASTERISK-19291: Background in realtime
Reported by: Andrew Nowrot

Category: PBX/pbx_spool

ASTERISK-17067: Long lines in call files cause spurious syntax error
Reported by: Dave OlszewskiASTERISK-17069: Callfile retries behave erratically as file size grows
Reported by: Jeremy Kister

Category: Resources/General

ASTERISK-28301: Allow voicemail boxes to be subscribed to with a presence event package
Reported by: George JosephASTERISK-28045: configure script does not enforce libunbound2 version
Reported by: Samuel GalarneauASTERISK-27553: [patch] res_curl: Avoid error message on unload.
Reported by: Alexander TraudASTERISK-21399: RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
Reported by: Tzafrir CohenASTERISK-25584: [patch] format-attribute module: VP8 missing
Reported by: Alexander TraudASTERISK-25108: configure check for older unbound library
Reported by: John BigelowASTERISK-25441: Deadlock in res_sorcery_memory_cache.
Reported by: Richard MudgettASTERISK-25110: res_resolver_unbound.c compilation failure: SIGURG is undeclared in func unbound_resolver_stop
Reported by: John Bigelow

Category: Resources/res_agi

ASTERISK-27621: (null) string tailing after AsyncAGIEnd AMI event
Reported by: sungtae kimASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP argument
Reported by: Rusty NewtonASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable" command without args
Reported by: Antoine PitrouASTERISK-25662: Malformed AGI 520 Usage response
Reported by: Tony MountifieldASTERISK-25951: res_agi: run_agi eats frames it shouldn't
Reported by: George JosephASTERISK-26343: ASTERISK-25951 causes issues for callerid manipulation through agi
Reported by: Morten TryfossASTERISK-25593: fastagi: record file closed after sending result
Reported by: Kevin HarwellASTERISK-23390: NewExten Event with application AGI shows up before and after AGI runs
Reported by: Benjamin Keith FordASTERISK-24323: Bug in documentation AGI STREAM FILE CONTROL
Reported by: Martin CisárikASTERISK-24027: MixMonitor AMI action called during AGI execution from bridge feature causes channel to leave AGI has hung up
Reported by: Matt Jordan

Category: Resources/res_ari

ASTERISK-28314: ARI: API changed but "apiVersion" in rest-api\resources.json did not
Reported by: Stefan RepkeASTERISK-28106: Astricon Feedback: Unable to filter ARI events when GETting causes overload of events
Reported by: George JosephASTERISK-28104: AstriCon Feedback: Automatically create a 1 line dialplan context for stasis apps
Reported by: George JosephASTERISK-27801: Asterisk got stuck while enabling "ari set debug all on"
Reported by: shaurya jainASTERISK-27445: ARI: Updating a bridge gives wrong error message.
Reported by: Frank DurdenASTERISK-27372: ARI: Node ARI client broken in latest versions of 13 and 14
Reported by: Benjamin Keith FordASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists
Reported by: Ronald RaikesASTERISK-26767: ARI channelvars cause memory leak
Reported by: Sébastien DuthilASTERISK-25492: ARI: Path parameters are case sensitive
Reported by: Joshua C. ColpASTERISK-25941: chan_pjsip: Crash on an immediate SIP final response
Reported by: Javier Riveros ASTERISK-25964: Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
Reported by: Matt JordanASTERISK-25882: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)
Reported by: Richard MudgettASTERISK-25771: ARI:Crash - Attended transfers of channels into Stasis application.
Reported by: Javier Riveros ASTERISK-25683: res_ari: Asterisk fails to start if compiled with MALLOC_DEBUG
Reported by: yaron nahumASTERISK-25522: ARI: Crash when creating channel via ARI originate with requesting channel
Reported by: Matt JordanASTERISK-25325: ARI PUT reload chan_sip HTTP response 404
Reported by: Rodrigo Ramirez NorambuenaASTERISK-25181: ARI: Channels added to Stasis application during WebSocket creation don't receive a StasisStart event
Reported by: Matt JordanASTERISK-25091: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge
Reported by: Ilya TrikozASTERISK-24812: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
Reported by: Matt JordanASTERISK-24501: ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd
Reported by: Matt JordanASTERISK-24339: Swagger API Docs have incorrect basePath
Reported by: Bradley WatkinsASTERISK-24264: ARI: Adding a channel to a holding bridge automatically starts MOH
Reported by: Samuel GalarneauASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Reported by: Matt JordanASTERISK-24043: ARI /continue fails to actually continue into the dialplan
Reported by: Krandon BruseASTERISK-24134: ARI: GET /channels/{channel_id}/variable for channel in dialplan returns 409 conflict
Reported by: Matt JordanASTERISK-24138: dial: Call forwarding information presented through AMI/ARI is wrong
Reported by: Matt Jordan

Category: Resources/res_ari_applications

ASTERISK-28302: ARI: "Error destroying mutex" when listing all ARI applications
Reported by: Stefan Repke

Category: Resources/res_ari_bridges

ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI calls
Reported by: Daniele PallastrelliASTERISK-25091: Asterisk REST API - bridge.addChannel crash asterisk when calling channel hangup while adding to bridge
Reported by: Ilya TrikozASTERISK-24560: Creating a named ARI bridge twice causes a crash
Reported by: Kinsey MooreASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John BigelowASTERISK-24591: Stasis() side of an ARI originated channel cannot be Redirected
Reported by: Kinsey MooreASTERISK-24264: ARI: Adding a channel to a holding bridge automatically starts MOH
Reported by: Samuel Galarneau

Category: Resources/res_ari_channels

ASTERISK-28181: ari: Originating overwrites channel start time
Reported by: sungtae kimASTERISK-28169: ARI /channels/create handler causes core dump
Reported by: sungtae kimASTERISK-27067: res_ari_channels: channel_state_invalid always leaks snapshot reference.
Reported by: Marin OdrljinASTERISK-26070: ari/channels: Creating a local channel without an originator adds all audio formats to it's capabilities
Reported by: George JosephASTERISK-25522: ARI: Crash when creating channel via ARI originate with requesting channel
Reported by: Matt JordanASTERISK-24812: ARI: Creating channels through /channels resource always uses SLIN, which results in unneeded transcoding
Reported by: Matt JordanASTERISK-24677: ARI GET variable on channel provides unhelpful response on non-existent variable
Reported by: Joshua C. ColpASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John BigelowASTERISK-24591: Stasis() side of an ARI originated channel cannot be Redirected
Reported by: Kinsey Moore

Category: Resources/res_ari_events

ASTERISK-25308: ari: Websocket leak
Reported by: Joshua C. Colp

Category: Resources/res_ari_playbacks

ASTERISK-26341: ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
Reported by: Matt JordanASTERISK-24229: ARI: playback of sounds implicitly answers channel, preventing early media playback
Reported by: Matt Jordan

Category: Resources/res_ari_recordings

ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error
Reported by: Tim Morgan

Category: Resources/res_calendar

ASTERISK-27680: [patch] res_calendar: Specialized calendars depend on symbols of general calendar.
Reported by: Alexander TraudASTERISK-25524: module reload res_calendar.so does not reload everything in calendar.conf
Reported by: JesperASTERISK-25523: res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
Reported by: JesperASTERISK-26683: res_calendar: Calendars duplicated after module reload
Reported by: Martin TomecASTERISK-25679: res_calendar leaks scheduler.
Reported by: Corey Farrell

Category: Resources/res_calendar_caldav

ASTERISK-24588: res_calendar does not process CalDAV from Owncloud [fix included]
Reported by: Stefan Gofferje

Category: Resources/res_calendar_ews

ASTERISK-24325: res_calendar_ews: cannot be used with neon 0.30
Reported by: Tzafrir Cohen

Category: Resources/res_calendar_icalendar

ASTERISK-27296: [patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
Reported by: Benoît Dereck-TricotASTERISK-27174: res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
Reported by: Mark Thompson

Category: Resources/res_clialiases

ASTERISK-20281: "core set verbose" behaves strangely, can't alias it, cli.conf example broken
Reported by: Tim Ringenbach at Asteria Solutions Group

Category: Resources/res_config_curl

ASTERISK-24676: Security Vulnerability: URL request injection in libCURL (CVE-2014-8150)
Reported by: Matt Jordan

Category: Resources/res_config_ldap

ASTERISK-26580: [patch] Error during LDAP modify action when user unregisters
Reported by: Nicholas John Koch

Category: Resources/res_config_odbc

ASTERISK-28341: res_config_odbc eliminates empty custom (“@” prefix) variables
Reported by: Alexei GradinariASTERISK-28166: app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC
Reported by: MichaelASTERISK-27863: config/ast_destroy_realtime_fields: successful DELETE is treated as failed
Reported by: Alexei GradinariASTERISK-26263: SQL error when using realtime and registering extension / inserting into ps_contacts
Reported by: Jeppe Ryskov LarsenASTERISK-26172: res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows.
Reported by: Alexei GradinariASTERISK-24808: res_config_odbc: Improper escaping of backslashes occurs with MySQL
Reported by: Javier Acosta

Category: Resources/res_config_pgsql

ASTERISK-27576: [patch] res_config_pgsql: Avoid typecasting an int to unsigned char.
Reported by: Alexander TraudASTERISK-27283: Realtime config fail with PostgreSQL version before 9.1
Reported by: Rodrigo Ramirez NorambuenaASTERISK-25628: res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
Reported by: Dmitry WaginASTERISK-15858: [patch] Fix query with double backslash in string literals and stop log warnings
Reported by: Humberto FigueraASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql
Reported by: mdu113

Category: Resources/res_config_sqlite

ASTERISK-27671: Deprecate legacy modules
Reported by: Corey Farrell

Category: Resources/res_config_sqlite3

ASTERISK-26057: res_config_sqlite3 uses incorrect query - unnecessary escape
Reported by: StepanASTERISK-23457: SQlite3: Realtime queue loading fails after PRAGMA query result
Reported by: Scott Griepentrog

Category: Resources/res_corosync

ASTERISK-25370: res_corosync segfaults at startup with corosync version > 2.x
Reported by: mdu113ASTERISK-24998: res_corosync: res_corosync tries to load even if res_corosync.conf is missing
Reported by: George Joseph

Category: Resources/res_crypto

ASTERISK-27908: [patch] crypto.h: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-25673: res_crypto leaks CLI entries
Reported by: Corey FarrellASTERISK-24550: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
Reported by: Osaulenko Alexander

Category: Resources/res_fax

ASTERISK-27981: res_fax: Fax session leak with fax gatewaying
Reported by: pasandevASTERISK-27657: res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available)
Reported by: Jared HullASTERISK-27094: res_fax: Deadlock when using Local channels and fax gateway
Reported by: David BrillertASTERISK-27364: channel: Crash when fax gateway is in use with PJSIP
Reported by: Jared HullASTERISK-27236: Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
Reported by: Ross BeerASTERISK-26203: res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
Reported by: Etienne LessardASTERISK-22732: Deadlock potential in res_fax and CCSS with local channels.
Reported by: Richard MudgettASTERISK-26216: res_fax: Deadlock when detect fax while channel executing Playback
Reported by: Richard MudgettASTERISK-26214: Allow arbitrary time for fax detection to end on a channel
Reported by: Richard MudgettASTERISK-26141: res_fax: fax_v21_session_new leaks reference to v21_details
Reported by: Corey FarrellASTERISK-25982: [patch]res_fax/t38_gateway: Peer V.21 session is created on wrong channel
Reported by: Alexei GradinariASTERISK-22790: check_modem_rate() may return incorrect rate for V.27
Reported by: not hereASTERISK-23231: Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load
Reported by: David BrillertASTERISK-24955: res_fax: v.27ter support baud rate of 2400, which is disallowed in res_fax's check_modem_rate
Reported by: Matt JordanASTERISK-24457: res_fax: fax gateway frames leak
Reported by: Corey FarrellASTERISK-24392: res_fax: fax gateway sessions leak
Reported by: Corey FarrellASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE
Reported by: not hereASTERISK-24357: [fax] Out of bounds error in update_modem_bits
Reported by: Jeremy LainéASTERISK-24301: Security: Out of call MESSAGE requests processed via Message channel driver can crash Asterisk
Reported by: Matt Jordan

Category: Resources/res_fax_spandsp

ASTERISK-18923: res_fax_spandsp usage counter is wrong
Reported by: Grigoriy Puzankin

Category: Resources/res_format_attr_h264

ASTERISK-27959: [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
Reported by: David KuehlingASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
Reported by: John HarrisASTERISK-24616: Crash in res_format_attr_h264 due to invalid string copy
Reported by: Yura Kocyuba

Category: Resources/res_format_attr_opus

ASTERISK-26579: codec_opus: Recursiveness when parsing fmtp line
Reported by: Jørgen HASTERISK-25583: [patch] format-attribute module: RFC 7587 (Opus Codec)
Reported by: Alexander Traud

Category: Resources/res_hep

ASTERISK-26758: res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets
Reported by: Max NorbaASTERISK-26953: Asterisk crash if hep.conf have some missing parameters
Reported by: Joel VandalASTERISK-26717: Document the fact that Asterisk HEP support only works with the PJSIP channel driver
Reported by: Olivier KriefASTERISK-26096: res_hep: Crash when configuration file is missing
Reported by: Niklas LarssonASTERISK-24491: Memory leak in res_hep
Reported by: Zane ConkleASTERISK-24362: res_hep leaks reference to configuration
Reported by: Corey Farrell

Category: Resources/res_hep_pjsip

ASTERISK-26758: res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets
Reported by: Max NorbaASTERISK-26850: res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
Reported by: Max NorbaASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations
Reported by: Matt Jordan

Category: Resources/res_hep_rtcp

ASTERISK-25352: res_hep_rtcp correlation_id is different then res_hep
Reported by: Kevin Scott AdamsASTERISK-24489: Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1
Reported by: Gregory MalsackASTERISK-24498: Segmentation fault in res_hep_rtcp on attended transfer
Reported by: Beppo MazzucatoASTERISK-24236: res_hep_rtcp: Module incorrectly depends on pjsip
Reported by: Matt Jordan

Category: Resources/res_http_post

ASTERISK-27719: [patch] res_http_post: Enable GMime in NetBSD.
Reported by: Alexander TraudASTERISK-27454: res_http_post: Don't require GMIME_MAJOR_VERSION
Reported by: Joshua C. Colp

Category: Resources/res_http_websocket

ASTERISK-28257: res_http_websocket: PING / PONG opcodes break data reception
Reported by: Jeremy LainéASTERISK-28231: res_http_websocket: Not responding to Connection Close Frame (opcode 8)
Reported by: Jeremy LainéASTERISK-27557: [patch] clang 5.0: implicit conversion to char changes value to negative.
Reported by: Alexander TraudASTERISK-27363: res_http_websocket: Wrong LocalAddress (it is equal to RemoteAddress)
Reported by: Vasilii RoginASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-26842: Websocket becomes disconnected when trying to place call from browser
Reported by: Mark MichelsonASTERISK-24330: Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
Reported by: Marek CervenkaASTERISK-24972: Transport Layer Security (TLS) Protocol BEAST Vulnerability - Investigate vulnerability of HTTP server
Reported by: Alex A. WelzlASTERISK-24106: WebSockets Automatically decides what driver it will use
Reported by: Andrew NagyASTERISK-25312: res_http_websocket: Terminate connection on fatal cases
Reported by: Joshua C. ColpASTERISK-24963: ASAN: heap-use-after-free with PJSIP and WSS
Reported by: Badalian VyacheslavASTERISK-24566: Uninit buf in WS write
Reported by: Badalian VyacheslavASTERISK-24472: Asterisk Crash in OpenSSL when calling over WSS from JSSIP
Reported by: Badalian VyacheslavASTERISK-24480: res_http_websockets: Module reference decrease below zero
Reported by: Corey Farrell

Category: Resources/res_jabber

ASTERISK-24425: [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
Reported by: abelbeckASTERISK-14233: [patch] Buddies are always auto-registered when processing the roster
Reported by: Simon Arlott

Category: Resources/res_monitor

ASTERISK-28249: res_monitor: Segfault with Monitor(wav,file,i)
Reported by: Valentin VidićASTERISK-27671: Deprecate legacy modules
Reported by: Corey FarrellASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-27103: core: ast_safe_system command injection possible.
Reported by: Corey FarrellASTERISK-24573: [patch]Out of sync conversation recording when divided in multiple recordings
Reported by: Nuno Borges

Category: Resources/res_musiconhold

ASTERISK-28029: [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file
Reported by: Frederic LE FOLLASTERISK-27774: res_musiconhold: Music on hold restarts after every announcement
Reported by: lvlASTERISK-27232: When in queue on g722 with interruptions, music on hold can get stuck and no longer play
Reported by: Jens T.ASTERISK-25974: Unused realtime MOH classes not purged on 'moh reload'
Reported by: Sébastien CoutureASTERISK-26086: res_musiconhold: format option is not documented adequately
Reported by: Jens BürgerASTERISK-23996: No core dumps because of res_musiconhold chdir.
Reported by: Walter DoekesASTERISK-26353: res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
Reported by: Jonathan HarrisASTERISK-25687: res_musiconhold: Concurrent invocations of 'moh reload' cause a crash
Reported by: Sean BrightASTERISK-24019: When a Music On Hold stream starts it restarts at beginning of file.
Reported by: Jason RichardsASTERISK-22252: res_musiconhold cleanup - REF_DEBUG reload warnings and ref leaks
Reported by: Walter Doekes

Category: Resources/res_mwi_external_ami

ASTERISK-25117: res_mwi_external_ami: Fix manager action registrations.
Reported by: Corey Farrell

Category: Resources/res_odbc

ASTERISK-28166: app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC
Reported by: MichaelASTERISK-28277: database: Add some basic logging
Reported by: Joshua C. ColpASTERISK-28065: res_odbc: missing SQL error diagnostic
Reported by: Alexei GradinariASTERISK-27722: [patch] BuildSystem: Depend not implicitly but explicitly on external libraries.
Reported by: Alexander TraudASTERISK-26704: res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
Reported by: Anthony MessinaASTERISK-26389: res_odbc: Clean up pooling options
Reported by: Joshua C. ColpASTERISK-25984: res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
Reported by: József DudásASTERISK-26074: res_odbc: Deadlock within UnixODBC
Reported by: Ross BeerASTERISK-25938: res_odbc: MySQL/MariaDB statement LAST_INSERT_ID() always returns zero.
Reported by: Edwin VandammeASTERISK-22708: res_odbc.conf negative_connection_cache option not respected, failover between DSNs doesn't work
Reported by: JoshEASTERISK-24742: [patch] Fix ast_odbc_find_table function in res_odbc
Reported by: ibercom

Category: Resources/res_parking

ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the same
Reported by: Ross BeerASTERISK-28616: parking: Deadlock when multi call parking
Reported by: Joshua C. ColpASTERISK-28173: Deadlock in chan_sip handling subscribe request during res_parking reload
Reported by: Giuseppe SucameliASTERISK-26399: app_queue: Agent not called when caller is parked
Reported by: wushumastersASTERISK-24605: res_parking option parkeddynamic does not work with the core Features 'parkcall' (DTMF initiated parking)
Reported by: Philip CorreiaASTERISK-24596: Unclear how to use Park application with res_parking 'parkeddynamic' enabled. Documentation?
Reported by: Philip CorreiaASTERISK-25369: res_parking: ParkAndAnnounce - Inheritable variables aren't applied to the announcer channel
Reported by: Jonathan RoseASTERISK-25254: Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park.
Reported by: Richard MudgettASTERISK-24899: Parking fall-through behavior different in 13
Reported by: Malcolm DavenportASTERISK-23850: Park Application does not respect Return Context Priority
Reported by: Andrew NagyASTERISK-24413: parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge
Reported by: Matt Jordan

Category: Resources/res_phoneprov

ASTERISK-26119: [patch] fix: memory leaks, resource leaks, out of bounds and bugs
Reported by: Alexei GradinariASTERISK-25721: [patch] res_phoneprov: memory leak and heap-use-after-free
Reported by: Badalian Vyacheslav

Category: Resources/res_pjsip

ASTERISK-28309: res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces
Reported by: Nikolay shakinASTERISK-28077: res_pjsip: improve realtime performance on CLI 'pjsip show contacts'
Reported by: Alexei GradinariASTERISK-27988: alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is integer not boolean
Reported by: Joshua C. ColpASTERISK-28022: res_pjsip realtime: uri column in ps_contacts table can be too short
Reported by: Florian FloimairASTERISK-27978: res_pjsip: Change default transport keepalive to preserve behavior
Reported by: Joshua C. ColpASTERISK-26686: res_pjsip: Lock inversion in transport management
Reported by: Ross BeerASTERISK-27872: res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload
Reported by: Alexei GradinariASTERISK-26806: pjsip_options: rework to make more efficient
Reported by: Kevin HarwellASTERISK-27688: res_pjsip: Crash on TCP PJSIP Transport Disconnect
Reported by: Ross BeerASTERISK-27679: res_pjsip: Endpoint destruction does not free DTLS configuration
Reported by: Mak DeeASTERISK-27571: res_pjsip: If SIP response is received during shutdown a crash may occur
Reported by: Joshua C. ColpASTERISK-25079: AMI bridge of channels results in MOH not destroyed and robotic audio on one channel
Reported by: Zane ConkleASTERISK-27345: res_pjsip_session: RTP instances leak on 488 responses.
Reported by: Corey FarrellASTERISK-27393: res_pjsip: Crash occurs when an empty contact read from astdb or database
Reported by: Aaron AnASTERISK-27032: res_pjsip: TLS options do not handle empty values
Reported by: seanchann.zhouASTERISK-27395: srtp: Add support for ephemeral DTLS certificates
Reported by: Sean BrightASTERISK-27387: Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any more
Reported by: Michael MaierASTERISK-27374: alembic: PJSIP scripts are missing column bundle in ps_endpoints table
Reported by: Florian FloimairASTERISK-27198: res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to yes
Reported by: Martin CisárikASTERISK-27047: res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
Reported by: dtrybaASTERISK-27254: alembic: prune_on_boot fix erroneous
Reported by: Florian FloimairASTERISK-26879: PJSIP external_media_address ignored if no local_net options are provided
Reported by: Matt JordanASTERISK-27168: alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
Reported by: Florian FloimairASTERISK-27147: Either asterisk or pjproject isn't re-using tcp connections (again)
Reported by: George JosephASTERISK-27119: res_pjsip: parse/add msid attribute when webrtc is enabled
Reported by: Kevin HarwellASTERISK-27090: PJSIP: Deadlock using TCP transport
Reported by: Richard MudgettASTERISK-26908: res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
Reported by: Richard MudgettASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
Reported by: Andreas KrügerASTERISK-26928: pjsip: Add database tables for PUBLISH support
Reported by: Joshua C. ColpASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross BeerASTERISK-26916: res_pjsip: Excessive refcount reached on transport ao2 object
Reported by: Ross BeerASTERISK-26363: res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
Reported by: Yaacov Akiba SlamaASTERISK-26685: res_pjsip: Crash when using IPv6 and Transport ws,wss
Reported by: Michael BalenASTERISK-26623: res_pjsip: Crash when calling PJSIPShowEndpoint
Reported by: Jørgen HASTERISK-26782: res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
Reported by: Peter SokolovASTERISK-26799: res_pjsip: Using an auth object for inbound and outbound authentication fails.
Reported by: Richard MudgettASTERISK-26738: Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
Reported by: Michael MaierASTERISK-26248: chan_pjsip: Error when calling PJSIP client with domain specified
Reported by: Norbert VargaASTERISK-26679: Crash on invalid contact domain (pjsip aor)
Reported by: DmitriyASTERISK-26699: res_pjsip: Assertion when sending OPTIONS request to endpoint
Reported by: Ross BeerASTERISK-26743: PJPROJECT: Detecting compiled max log level does not work.
Reported by: Richard MudgettASTERISK-26684: res_pjsip: Various issues with compact SIP headers
Reported by: Joshua ElsonASTERISK-24499: Need more explicit debug when PJSIP dialstring is invalid
Reported by: Rusty NewtonASTERISK-26490: res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
Reported by: Juris BreicisASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard MudgettASTERISK-26571: res_pjsip: Resolution incorrect when explicit IPv6 transport configured
Reported by: Joshua C. ColpASTERISK-26453: res_pjsip_config_wizard: Memory leak in module_unload
Reported by: Badalian VyacheslavASTERISK-26375: res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
Reported by: Joshua C. ColpASTERISK-26364: res_pjsip: Don't assume a request will have target addresses
Reported by: Joshua C. ColpASTERISK-26264: res_pjsip: Crash when applying ACL from non-existent endpoint
Reported by: nappsoftASTERISK-26319: [patch] res_pjsip: qualify/unqualify added/deleted realtime endpoints
Reported by: Alexei GradinariASTERISK-26269: res_pjsip: Wrong state for aors without registered contacts after startup
Reported by: nappsoftASTERISK-22374: Finish mapping the sip.conf parameters to res_sip.conf parameters
Reported by: Matt JordanASTERISK-26305: Asterisk 14: Two resolver unbound testsuite tests fail
Reported by: Richard MudgettASTERISK-26241: res_pjsip: When using compact headers, rpid and pai are incorrectly generated
Reported by: George JosephASTERISK-26238: res_pjsip: Empty global default_from_user causes crash
Reported by: Joshua C. ColpASTERISK-26145: pjsip: Deadlock with suspend + masquerade + indicate
Reported by: Ross BeerASTERISK-26206: [patch] res_pjsip: Use more compatible regex for get all
Reported by: Dmitry WaginASTERISK-26256: [patch] SIP/SDP origin (o=) contains brackets with IP6
Reported by: Alexander TraudASTERISK-26174: res_pjsip: Crash when freeing cloned message in distributor
Reported by: Ross BeerASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code.
Reported by: Corey FarrellASTERISK-26160: pjsip: Updated->Reachable during qualify
Reported by: Matt JordanASTERISK-25772: res_pjsip: Unexpected two BYE when answered
Reported by: Dmitriy SerovASTERISK-26061: [patch] res_pjsip: improve realtime performance - remove updating all endpoints status on startup
Reported by: Alexei GradinariASTERISK-26049: res_pjsip: Crash when our own request timer fires
Reported by: Joshua C. ColpASTERISK-25941: chan_pjsip: Crash on an immediate SIP final response
Reported by: Javier Riveros ASTERISK-26007: res_pjsip: Endpoints deleting early after upgrade from 13.8.2 to 13.9
Reported by: Greg SiemonASTERISK-26004: res_pjsip: The transport/method parameter is ignored
Reported by: George JosephASTERISK-25928: res_pjsip: URI validation done outside of PJSIP thread
Reported by: Joshua C. ColpASTERISK-25914: PJSIP: failed registration with wrong codec name on allow/disallow
Reported by: Alexei GradinariASTERISK-25796: res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES
Reported by: George JosephASTERISK-25707: Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions
Reported by: George JosephASTERISK-25123: Bracketed IPv6 Contact header parameter unparsable with Asterisk/PJSIP
Reported by: Anthony MessinaASTERISK-25885: res_pjsip: Race condition between adding contact and automatic expiration
Reported by: Joshua C. ColpASTERISK-25829: res_pjsip: PJSIP does not accept spaces when separating multiple AORs
Reported by: Mateusz KowalskiASTERISK-25727: RPM build requires OPTIONAL_API cflag due to PJSIP requirement
Reported by: Gergely DömsödiASTERISK-25337: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
Reported by: Jacques PeacockASTERISK-25751: res_pjsip: Support pjsip_dlg_create_uas_and_inc_lock
Reported by: Joshua C. ColpASTERISK-25606: Core dump when using transports in sorcery
Reported by: Martin MoučkaASTERISK-25702: PjSip realtime DB and Cache Errors since upgrade to asterisk-13.7.0 from asterisk-13.7.0-rc2
Reported by: Nic ColledgeASTERISK-25712: Second call to already-on-call phone and Asterisk sends "Ready"
Reported by: Richard MudgettASTERISK-25686: PJSIP: qualify_timeout is a double, database schema is an integer
Reported by: Marcelo TerresASTERISK-25668: res_pjsip: Deadlock in distributor
Reported by: Mark MichelsonASTERISK-25116: res_pjsip: Two PeerStatus AMI messages are sent for every status change
Reported by: George JosephASTERISK-25608: res_pjsip/contacts/statsd: Lifecycle events aren't consistent
Reported by: George JosephASTERISK-25595: Unescaped : in messge sent to statsd
Reported by: Niklas LarssonASTERISK-25598: res_pjsip: Contact status messages are printing a hash instead of the uri
Reported by: George JosephASTERISK-25486: res_pjsip: Fix deadlock when validating URIs
Reported by: Joshua C. ColpASTERISK-25455: Deadlock of PJSIP realtime over res_config_pgsql
Reported by: mdu113ASTERISK-25295: res_pjsip crash - pjsip_uri_get_uri at /usr/include/pjsip/sip_uri.h
Reported by: Dmitriy SerovASTERISK-25381: res_pjsip: AoRs deleted via ARI (or other mechanism) do not destroy their related contacts
Reported by: Matt JordanASTERISK-25339: res_pjsip: Empty "auth" sections from non-config backgrounds are interpreted as valid
Reported by: Matt JordanASTERISK-25304: res_pjsip: XML sanitization may write past buffer
Reported by: Joshua C. ColpASTERISK-25201: Crash in PJSIP distributor on already free'd threadpool
Reported by: Matt JordanASTERISK-25168: Random Core Dumps on Asterisk 13.4 PJSIP, in ast_channel_name at channel_internal_api.c
Reported by: Carl FortinASTERISK-25076: res_pjsip: Failover does not occur on connection-less transport or 503 response
Reported by: Joshua C. ColpASTERISK-25171: Early completion of feature code attended transfer results in intermittent one-way audio, "ghost ringing" and robotic sound.
Reported by: Rusty NewtonASTERISK-25158: res_pjsip: Add option to use AAL2 packing when negotiating g.726
Reported by: Kevin HarwellASTERISK-25115: Crash related to func sip_resolve_invoke_user_callback of res_pjsip/pjsip_resolver.c
Reported by: John BigelowASTERISK-25096: [patch]Segfault when registering over websockets with PJSIP (in ast_sockaddr_isnull at /include/asterisk/netsock2.h)
Reported by: Josh KitchensASTERISK-25131: chan_pjsip: In-dialog authentication not handled.
Reported by: Richard MudgettASTERISK-25105: res_pjsip: Possible incompatibility between qualify_timeout and pjproject-2.4
Reported by: George JosephASTERISK-25089: res_pjsip_config_wizard: Variable specified in templates aren't being processed correctly
Reported by: George JosephASTERISK-25033: Asterisk 13 (branch head) won't compile without PJSip
Reported by: Peter WhiskerASTERISK-25020: Mismatched response to outgoing REGISTER request
Reported by: Mark MichelsonASTERISK-24999: PJSIP crashes with malformed contact line
Reported by: snuffyASTERISK-24977: Contacts that don't use qualify are being marked as unavailable
Reported by: George JosephASTERISK-24863: res_pjsip: No endpoint events raised via AMI when contacts cannot be reached/qualified
Reported by: Dmitriy SerovASTERISK-24380: core: Native formats are set to h264 with certain audio/video codec configuration, resulting in path translation WARNINGs
Reported by: Matt JordanASTERISK-24935: res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator.
Reported by: Corey FarrellASTERISK-24910: "timer=no" and "timer=required" settings in pjsip.conf fail
Reported by: Ray CrumrineASTERISK-24920: Asterisk handles duplicate SIP requests as if they were each a new request
Reported by: Mark MichelsonASTERISK-24840: res_pjsip: conflicting endpoint identifiers
Reported by: Kevin HarwellASTERISK-24872: [patch] AMI PJSIPShowEndpoint closes AMI connection on error
Reported by: Dmitriy SerovASTERISK-24755: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
Reported by: John BigelowASTERISK-24685: "pjsip show version" CLI command
Reported by: Joshua C. ColpASTERISK-24727: PJSIP: Crash experienced during multi-Asterisk transfer scenario.
Reported by: Mark MichelsonASTERISK-24741: dtls_handler causes Asterisk to crash
Reported by: Zane ConkleASTERISK-24748: res_pjsip: If wizards explicitly configured in sorcery.conf false ERROR messages may occur
Reported by: Joshua C. ColpASTERISK-24485: res_pjsip cannot be unloaded or shutdown
Reported by: Corey FarrellASTERISK-24615: When Multiple Transports Exist in pjsip.conf, Incorrect External Addresses is Used in SIP Packets When Responding to INVITE
Reported by: David JustlASTERISK-24367: PJSIP: allow all results in failure to send INVITE
Reported by: Scott GriepentrogASTERISK-24342: PJSIP: Qualifying endpoints attempts to do them all at the same time.
Reported by: Richard MudgettASTERISK-24471: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
Reported by: yaron nahumASTERISK-24508: pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"
Reported by: Beppo MazzucatoASTERISK-24336: PJSIP timer_min_se value under 90 causes crash
Reported by: Leon RowlandASTERISK-24462: res_pjsip: Stale qualify statistics after disablementation
Reported by: Kevin HarwellASTERISK-24122: Documentaton for res_pjsip option use_avpf needs to be fixed
Reported by: James Van VleetASTERISK-24312: SIGABRT when improperly configured realtime pjsip
Reported by: Dafi NiASTERISK-24387: res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on
Reported by: Matt JordanASTERISK-24370: res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd
Reported by: Matt JordanASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations
Reported by: Matt JordanASTERISK-24199: 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid
Reported by: Joshua C. ColpASTERISK-24350: PJSIP shows commands prints unneeded headers
Reported by: snuffyASTERISK-24295: crash: creating out of dialog OPTIONS request crashes
Reported by: Rogger PadillaASTERISK-24161: PJSIPShowEndpoint gives inaccurate count of list items
Reported by: Mark Michelson

Category: Resources/res_pjsip/Bundling

ASTERISK-28059: PJSIP: Update bundled PJPROJECT to version 2.8
Reported by: Joshua C. ColpASTERISK-26980: pjsip: Clean up WebRTC disables
Reported by: abelbeckASTERISK-27411: pjsip: TCP connections may not be destroyed
Reported by: Joshua C. ColpASTERISK-27052: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
Reported by: alexASTERISK-26927: pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
Reported by: Alexander TraudASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross BeerASTERISK-26743: PJPROJECT: Detecting compiled max log level does not work.
Reported by: Richard MudgettASTERISK-26416: pjproject-bundled: configure fails to check for all required utilities
Reported by: Corey FarrellASTERISK-26148: pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
Reported by: Hans van EijsdenASTERISK-25873: res_pjsip: Bundled pjproject: compile error, cannot find -lasteriskpj
Reported by: Hans van Eijsden

Category: Resources/res_pjsip_acl

ASTERISK-24531: res_pjsip_acl: ACLs not applied on initial module load
Reported by: Matt Jordan

Category: Resources/res_pjsip_authenticator_digest

ASTERISK-26799: res_pjsip: Using an auth object for inbound and outbound authentication fails.
Reported by: Richard Mudgett

Category: Resources/res_pjsip_caller_id

ASTERISK-27284: Status of RFC 3323 and PJSIP
Reported by: dtrybaASTERISK-25823: SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
Reported by: Andreas KrügerASTERISK-26307: res_pjsip_caller_id: Crash on outgoing change
Reported by: Bill BrigdenASTERISK-26316: res_pjsip_callerid: Irregular URI causes unexpected callerid
Reported by: Kevin HarwellASTERISK-25942: res_pjsip_caller_id: Transfer results in mixed ConnectedLine information
Reported by: George Joseph

Category: Resources/res_pjsip_config_wizard

ASTERISK-27992: PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash
Reported by: Jonathan Harris

Category: Resources/res_pjsip_dialog_info_body_generator

ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip
Reported by: Zach RASTERISK-25999: res_pjsip_dialog_info_body_generator: Remove subscription requirement
Reported by: Joshua C. Colp

Category: Resources/res_pjsip_diversion

ASTERISK-28312: res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect
Reported by: Alex Odrov

Category: Resources/res_pjsip_endpoint_identifier_ip

ASTERISK-27548: res_pjsip_endpoint_identifier_ip only matches against "generic string" headers
Reported by: George JosephASTERISK-27861: [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers.
Reported by: Alexander TraudASTERISK-27491: res_pjsip_endpoint_identifier_ip only matches against header if match by ip fails
Reported by: George JosephASTERISK-26735: res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
Reported by: Michael MaierASTERISK-26693: res_pjsip_endpoint_identifier_ip: Add support for SRV
Reported by: Joshua C. ColpASTERISK-24290: Endpoint identifier match value fails to parse when CIDR network format is specified
Reported by: Ray Crumrine

Category: Resources/res_pjsip_exten_state

ASTERISK-25922: res_pjsip_exten_state: Add configuration support for publishing
Reported by: Joshua C. ColpASTERISK-24716: Improve pjsip log messages for presence subscription failure
Reported by: Rusty Newton

Category: Resources/res_pjsip_keepalive

ASTERISK-27347: [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
Reported by: Alexander Traud

Category: Resources/res_pjsip_logger

ASTERISK-26239: res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
Reported by: George JosephASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations
Reported by: Matt Jordan

Category: Resources/res_pjsip_messaging

ASTERISK-27942: res_pjsip_messaging doesn't accept application/* content-types.
Reported by: George JosephASTERISK-27193: IPv6 receive address in message doesn't include brackets
Reported by: Scott GriepentrogASTERISK-26484: res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
Reported by: Vinod DharashiveASTERISK-24937: [patch]res_pjsip_messaging: Messages may be sent out of order
Reported by: Mark Michelson

Category: Resources/res_pjsip_multihomed

ASTERISK-26374: res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
Reported by: Joshua C. ColpASTERISK-24438: res_pjsip_multihomed.so blocks Asterisk reload when DNS settings invalid
Reported by: Melissa Shepherd

Category: Resources/res_pjsip_mwi

ASTERISK-28306: res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent
Reported by: Jared HullASTERISK-27121: res_pjsip_mwi: Memory leak on reload
Reported by: Sergej KasumovicASTERISK-27652: Null pointer Crash in PJSIP MWI
Reported by: Joshua ElsonASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact
Reported by: Alexei GradinariASTERISK-26756: res_pjsip_mwi: Asterisk does not terminate MWI subscription
Reported by: Carl FortinASTERISK-26200: [patch] res_pjsip_mwi: improve realtime performance - remove unneeded check on endpoint's contacts.
Reported by: Alexei GradinariASTERISK-26065: chan_pjsip: MWI NOTIFY contents not ordered properly
Reported by: Ross BeerASTERISK-25180: res_pjsip_mwi: Unsolicited MWI requires reload
Reported by: Joshua C. ColpASTERISK-24982: res_pjsip_mwi: Unsolicited MWI NOTIFY only sent on mailbox changes
Reported by: Joshua C. Colp

Category: Resources/res_pjsip_mwi_body_generator

ASTERISK-26065: chan_pjsip: MWI NOTIFY contents not ordered properly
Reported by: Ross Beer

Category: Resources/res_pjsip_nat

ASTERISK-28129: Incorrect Behavior for rewrite_contact when Re-Invite omits routset
Reported by: Torrey SearleASTERISK-25830: Revision 2451d4e breaks NAT
Reported by: Sean BrightASTERISK-25387: res_pjsip_nat: Malformed REGISTER request causes NAT'd Contact header to not be rewritten
Reported by: Matt JordanASTERISK-25196: res_pjsip_nat: rewrite_contact should not be applied to Contact header when Record-Route headers are present
Reported by: Mark MichelsonASTERISK-23634: With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls
Reported by: Roman Skvirsky

Category: Resources/res_pjsip_notify

ASTERISK-28137: res_pjsip_notify: improve realtime performance on CLI completion on the endpoint
Reported by: Alexei GradinariASTERISK-25590: CLI Usage info for 'pjsip send notify' references incorrect config
Reported by: Corey Farrell

Category: Resources/res_pjsip_outbound_publish

ASTERISK-27298: Problem with expires on pjsip / outbound-publish
Reported by: Cyrille DemaretASTERISK-26506: [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
Reported by: Matt KrokoszASTERISK-25217: [patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
Reported by: Richard MudgettASTERISK-26053: res_pjsip_outbound_publish: Crash when shutting down
Reported by: Joshua C. Colp

Category: Resources/res_pjsip_outbound_registration

ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
Reported by: Kevin HarwellASTERISK-26808: res_pjsip_outbound_registration doesn't know about network change events
Reported by: George JosephASTERISK-26782: res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
Reported by: Peter SokolovASTERISK-25964: Outbound registrations created via ARI/push configuration do not clean up outbound registrations currently in flight
Reported by: Matt JordanASTERISK-25990: PJSIP TLS registration should respect client_uri scheme when generating Contact URI
Reported by: Sebastian DammASTERISK-25737: res_pjsip_outbound_registration: line option not in Alembic
Reported by: Joshua C. ColpASTERISK-25575: res_pjsip: Dynamic outbound registrations created via ARI are not loaded into memory on Asterisk start/restart
Reported by: Matt JordanASTERISK-25485: res_pjsip_outbound_registration: registration stops due to 400 response
Reported by: Kevin HarwellASTERISK-24907: res_pjsip_outbound_registration: crash during unload if registration attempts are still occuring
Reported by: Kevin HarwellASTERISK-25037: res_pjsip_outbound_registration: Potential crash in off-nominal failure case when sending message
Reported by: Joshua C. ColpASTERISK-24729: Outbound registration not occuring on new registrations after reload.
Reported by: Richard MudgettASTERISK-24514: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard
Reported by: Kevin HarwellASTERISK-24411: [patch] Status of outbound registration is not changed upon unregistering.
Reported by: John BigelowASTERISK-24398: Initialize auth_rejection_permanent on client state to the configuration parameter value
Reported by: Matt Jordan

Category: Resources/res_pjsip_pidf_body_generator

ASTERISK-27290: res_pjsip: PIDF contact field has malformed/invalid XML
Reported by: basildane

Category: Resources/res_pjsip_pidf_eyebeam_body_supplement

ASTERISK-26659: res_pjsip: PJSIP presence - missing braces around the status element in XML
Reported by: Abraham Liebsch

Category: Resources/res_pjsip_publish_asterisk

ASTERISK-24635: PJSIP outbound PUBLISH crashes when no response is ever received
Reported by: Marco Paland

Category: Resources/res_pjsip_pubsub

ASTERISK-27956: res_pjsip_pubsub: segfault in function publish_expire
Reported by: Alexei GradinariASTERISK-27783: res_pjsip_pubsub: apparent crash on shutdown
Reported by: Kevin HarwellASTERISK-27612: Subscriptions Persist After Expiration and TCP/TLS Disconnect
Reported by: Ross BeerASTERISK-24483: res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1
Reported by: Tzafrir CohenASTERISK-27279: Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
Reported by: Ross BeerASTERISK-26929: pjsip: Add database tables for RLS
Reported by: Joshua C. ColpASTERISK-26776: res_pjsip_pubsub: Crash when generating xpidf content
Reported by: Andrew GreenASTERISK-26823: PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
Reported by: Mark MichelsonASTERISK-26696: pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
Reported by: Zach RASTERISK-26164: XMPP no longer triggers NOTIFY to device via chan_pjsip
Reported by: Ross BeerASTERISK-26166: res_pjsip_pubsub: Crash when decrementing reference count of message
Reported by: Ross BeerASTERISK-26099: res_pjsip_pubsub: Crash when sending request due to server timeout
Reported by: Ross BeerASTERISK-25738: res_pjsip_pubsub: Crash while executing OutboundSubscriptionDetail ami action
Reported by: Kevin HarwellASTERISK-25513: Crash: malloc failed with high load of subscriptions.
Reported by: John BigelowASTERISK-25505: res_pjsip_pubsub: Crash on off-nominal when UAS dialog can't be created
Reported by: Joshua C. ColpASTERISK-25306: Persistent subscriptions can save multiple SIP messages at once, leading to potential crashes.
Reported by: Mark MichelsonASTERISK-25057: res_pjsip_pubsub: Crash in send_notify due to invalid root pointer in sub_tree
Reported by: Matt JordanASTERISK-24970: Crash in res_pjsip_pubsub handling of failed notify
Reported by: Scott GriepentrogASTERISK-24368: res_pjsip_pubsub: Subscription persistence causes crash when re-constructing stored subscription
Reported by: Matt JordanASTERISK-24136: Security: Crash in Asterisk's PJSIP code when subscribing to an event with an unexpected body type
Reported by: Mark MichelsonASTERISK-24181: RLS: Large lists don't get sent because they exceed the PJSIP message length limit
Reported by: Jonathan Rose

Category: Resources/res_pjsip_refer

ASTERISK-27568: PJSIP: Crash during SIP attended transfer.
Reported by: Bryan WaltersASTERISK-24483: res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == -1
Reported by: Tzafrir CohenASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin HarwellASTERISK-26869: res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
Reported by: Torrey SearleASTERISK-25814: Segfault at f ip in res_pjsip_refer.so
Reported by: Sergio Medina ToledoASTERISK-25204: res_pjsip_refer: Duplicated Referred-By or Replaces headers on outbound INVITEs.
Reported by: Mark MichelsonASTERISK-24700: CRASH: NULL channel is being passed to ast_bridge_transfer_attended()
Reported by: Zane ConkleASTERISK-24376: 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
Reported by: Matt JordanASTERISK-24528: res_pjsip_refer: Sending INVITE with Replaces in-dialog with invalid target causes crash
Reported by: Joshua C. ColpASTERISK-24508: pjsip - REFER request from SNOM is rejected with "400 bad request" - DEBUG shows "Received a REFER without a parseable Refer-To"
Reported by: Beppo Mazzucato

Category: Resources/res_pjsip_registrar

ASTERISK-28001: res_pjsip_registrar: Improve performance of inbound handling
Reported by: Joshua C. ColpASTERISK-27192: res_pjsip: Loss of SIP registrations causing unavailable endpoints
Reported by: Richard MudgettASTERISK-26644: PJSIPShowRegistrationsInbound just dumps all aors
Reported by: George JosephASTERISK-25929: res_pjsip_registrar: AOR_CONTACT_ADDED events not raised
Reported by: Joshua C. ColpASTERISK-25885: res_pjsip: Race condition between adding contact and automatic expiration
Reported by: Joshua C. ColpASTERISK-24785: 'Expires' header missing from 200 OK on REGISTER
Reported by: Ross Beer

Category: Resources/res_pjsip_rfc3326

ASTERISK-27949: res_pjsip_rfc3326: A lot of endpoints do not correctly handle two Reason headers
Reported by: Ross BeerASTERISK-27741: res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't account for more than one 'Reason' header
Reported by: Ross BeerASTERISK-27554: res_pjsip_rfc3326: Order of 'Reason' headers break many endpoints
Reported by: Ross Beer

Category: Resources/res_pjsip_sdp_rtp

ASTERISK-28110: rtp: Incorrect Packetization
Reported by: Robert CrippsASTERISK-28007: rtcp-mux is put in SDP answer regardless of offer
Reported by: Torrey SearleASTERISK-27398: No joint capabilities with video and audio-only streams
Reported by: Benjamin Keith FordASTERISK-27957: PJSIP proposes ICE candidates on answer even if not in offer
Reported by: Torrey SearleASTERISK-27345: res_pjsip_session: RTP instances leak on 488 responses.
Reported by: Corey FarrellASTERISK-27179: res_pjsip_session: Handling of 'msid' is incorrect
Reported by: Kevin HarwellASTERISK-26890: STUN server with non-default-route transport causes INVITE delay
Reported by: George JosephASTERISK-26851: res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
Reported by: Richard BeggASTERISK-26541: res_pjsip_sdp_rtp: Restrict number of formats to maximum
Reported by: Joshua C. ColpASTERISK-26423: res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
Reported by: Andreas WetzelASTERISK-26309: [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
Reported by: Alexander TraudASTERISK-26228: res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
Reported by: Ali GhavidelASTERISK-26119: [patch] fix: memory leaks, resource leaks, out of bounds and bugs
Reported by: Alexei GradinariASTERISK-25854: No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk
Reported by: Robert McGilvrayASTERISK-25632: res_pjsip_sdp_rtp: RTP is sent from wrong IP address when multihomed
Reported by: Olivier KriefASTERISK-25356: res_pjsip_sdp_rtp: Multiple keepalive scheduled items may exist
Reported by: Joshua C. ColpASTERISK-24769: res_pjsip_sdp_rtp: Local ICE candidates leaked
Reported by: Matt JordanASTERISK-24381: res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections
Reported by: Matt JordanASTERISK-24222: PJSIP: Failed assertions when placing a call with no allow= specified
Reported by: Mark MichelsonASTERISK-23994: res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname
Reported by: Private Name

Category: Resources/res_pjsip_session

ASTERISK-28157: Asterisk crashes when the res_pjsip_* modules unload
Reported by: sungtae kimASTERISK-28047: chan_pjsip: Declined video stream is added when no video codecs configured and session refresh with removed video stream occurs
Reported by: WillASTERISK-27955: res_pjsip_session: sdp group:BUNDLE attribute truncated
Reported by: Kevin HarwellASTERISK-27763: res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream
Reported by: Thiago CoutinhoASTERISK-27936: res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183
Reported by: George JosephASTERISK-27614: res_pjsip_session: SDP origin does not use resolved address
Reported by: John M.ASTERISK-27566: res_pjsip_session: Improve WebRTC interop with bundling during renegotiation
Reported by: Joshua C. ColpASTERISK-27345: res_pjsip_session: RTP instances leak on 488 responses.
Reported by: Corey FarrellASTERISK-27341: [patch] res_pjsip_session: SIP/SDP origin (o=) contains local address.
Reported by: Alexander TraudASTERISK-26988: res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
Reported by: dtrybaASTERISK-27264: res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
Reported by: Daniel HecklASTERISK-27024: nat/external_media settings ignored in 14.4.1
Reported by: Christopher van de SandeASTERISK-27209: Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
Reported by: Torrey SearleASTERISK-27110: RTP session is not fully destroyed on channel hangup
Reported by: Matt JordanASTERISK-27179: res_pjsip_session: Handling of 'msid' is incorrect
Reported by: Kevin HarwellASTERISK-27143: bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
Reported by: Joshua C. ColpASTERISK-27118: res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
Reported by: Joshua C. ColpASTERISK-27076: chan_pjsip: Add support for multiple streams
Reported by: Joshua C. ColpASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer
Reported by: Kevin HarwellASTERISK-26998: res_pjsip_session: INVITE retransmissions could still setup the same call again.
Reported by: Richard MudgettASTERISK-26908: res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
Reported by: Richard MudgettASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and To URI differ
Reported by: Yasin CANERASTERISK-26670: [patch] Outgoing SIP-URI Dialing via PJSIP
Reported by: Alexander TraudASTERISK-26317: res_pjsip_session: Add ability to use preferred codec only
Reported by: Aaron AnASTERISK-26291: res_pjsip_session: segfault on already disconnected session
Reported by: Alexei GradinariASTERISK-26127: res_pjsip_session: Crash due to race condition between res_pjsip_session unload and timer
Reported by: Joshua C. ColpASTERISK-25297: Crashes running channels/pjsip/resolver/srv/failover/in_dialog testsuite tests
Reported by: Richard MudgettASTERISK-25131: chan_pjsip: In-dialog authentication not handled.
Reported by: Richard MudgettASTERISK-25086: [patch]PJSIP crashes if endpoint missing in Dial()
Reported by: snuffyASTERISK-24731: res_pjsip_session cannot be unloaded
Reported by: Corey FarrellASTERISK-24607: res_pjsip_session: re-INVITE with declined media streams results in 488
Reported by: Matt Jordan

Category: Resources/res_pjsip_t38

ASTERISK-27944: res_pjsip_t38: Crash receiving 1xx responses other than 100 before 200 for T.38 reINVITE
Reported by: Joshua ElsonASTERISK-27657: res_pjsip_t38: ATA fails with hangupcause 58(Bearer capability not available)
Reported by: Jared HullASTERISK-27080: res_pjsip_t38: Slow T.38 re-invite rejection if remote leg has T.38 disabled
Reported by: Torrey SearleASTERISK-27364: channel: Crash when fax gateway is in use with PJSIP
Reported by: Jared HullASTERISK-27236: Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
Reported by: Ross BeerASTERISK-26974: res_pjsip: Deadlock in T.38 framehook
Reported by: Richard MudgettASTERISK-25582: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
Reported by: Matt JordanASTERISK-24928: [patch]t38_udptl_maxdatagram in pjsip.conf not honored
Reported by: Juergen SpiesASTERISK-24933: T38 fails negotiation
Reported by: Jonathan Rose

Category: Resources/res_pjsip_transport_websocket

ASTERISK-28020: res_pjsip_transport_websocket: Properly set 'received' for IPv6
Reported by: Sean BrightASTERISK-27046: res_pjsip_transport_websocket: segfault in get_write_timeout
Reported by: Jørgen HASTERISK-26796: res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
Reported by: Jørgen HASTERISK-24106: WebSockets Automatically decides what driver it will use
Reported by: Andrew NagyASTERISK-25122: Large SIP packet received via pjsip over websocket crashes Asterisk
Reported by: Ivan PoddubnyASTERISK-24143: pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
Reported by: Aleksei Kulakov

Category: Resources/res_realtime

ASTERISK-26172: res_sorcery_realtime: fix bug when successful sql UPDATE is treated as failed if there is no affected rows.
Reported by: Alexei GradinariASTERISK-25914: PJSIP: failed registration with wrong codec name on allow/disallow
Reported by: Alexei Gradinari

Category: Resources/res_rtp_asterisk

ASTERISK-28321: res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation
Reported by: sungtae kimASTERISK-28303: res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps
Reported by: Torrey SearleASTERISK-28284: switching between native_bridge and simple_bridge can cause one way audio
Reported by: Torrey SearleASTERISK-28230: res_rtp_asterisk: abs-send-time extension added with Asterisk 15.5.0 breaks GXV3140 video telephony
Reported by: David KuehlingASTERISK-28162: [patch] need to reset DTMF last sequence number and timestamp on RTP renegotiation
Reported by: Alexei GradinariASTERISK-28110: rtp: Incorrect Packetization
Reported by: Robert CrippsASTERISK-28002: When T.140 realtime text is negociated, a lot of debug traces are generated
Reported by: Emmanuel BUUASTERISK-27990: res_rtp_asterisk: Requires OpenSSL in Developer Mode.
Reported by: Alexander TraudASTERISK-27810: BASIC-RETRANS: Implement receive
Reported by: Benjamin Keith FordASTERISK-27848: rtp: DTMF Breaks With telephony-event/16000
Reported by: DominicASTERISK-27845: Codec-Change Re-INVITE during DTMF can cause marker bit error
Reported by: Torrey SearleASTERISK-27831: res_rtp_asterisk: Add support for abs-send-time RTP extension
Reported by: Joshua C. ColpASTERISK-27806: BASIC-RETRANS: Implement send
Reported by: Benjamin Keith FordASTERISK-27776: res_rtp_asterisk: Add support for sending RTCP feedback messages
Reported by: Joshua C. ColpASTERISK-27758: res_rtp_asterisk: Add support for raising RTCP feedback messages
Reported by: Joshua C. ColpASTERISK-27440: Strictrtp has issues to qualify video rtp streams
Reported by: Wim De VlaminckASTERISK-27429: res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it should
Reported by: Vitezslav NovyASTERISK-27453: RTP: Blind transfer direct media scenario results in one way audio.
Reported by: Richard MudgettASTERISK-27437: [patch] ICE: server-reflexive candidates (srflx) with Dual-Stack.
Reported by: Alexander TraudASTERISK-27421: RTP source learning not working with devices that have some clock issues
Reported by: nappsoftASTERISK-27395: srtp: Add support for ephemeral DTLS certificates
Reported by: Sean BrightASTERISK-27328: Missing openssl dependencies in res_rtp_asterisk and tcptls
Reported by: Tzafrir CohenASTERISK-27292: Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
Reported by: Ross BeerASTERISK-27274: RTCP needs better packet validation to resist port scans.
Reported by: Richard MudgettASTERISK-27252: RTP: One way audio with direct media and strictrtp=yes.
Reported by: Richard MudgettASTERISK-27013: res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
Reported by: Joshua C. ColpASTERISK-27231: res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
Reported by: Joshua C. ColpASTERISK-27158: [patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
Reported by: Torrey SearleASTERISK-27143: bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
Reported by: Joshua C. ColpASTERISK-27133: res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
Reported by: Joshua C. ColpASTERISK-27118: res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
Reported by: Joshua C. ColpASTERISK-27023: res_rtp_asterisk: Deadlock when TURN session in use
Reported by: Jatin JainASTERISK-27096: res_rtp_asterisk: add a control frame for when dtls is established
Reported by: Kevin HarwellASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component
Reported by: Michael WaltonASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
Reported by: Frankie ChinASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
Reported by: Javier Riveros ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
Reported by: Stefan EngströmASTERISK-26143: res_rtp_asterisk: One way audio when transcoding
Reported by: Henning HoltschneiderASTERISK-26692: res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
Reported by: Sebastian GutierrezASTERISK-26835: res_rtp_asterisk: Crash when freeing RTCP address string
Reported by: Niklas LarssonASTERISK-26853: res_rtp_asterisk: Crash in pjnath when receiving packet
Reported by: AdagioASTERISK-26732: res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
Reported by: Dan JenkinsASTERISK-26710: [patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
Reported by: Aaron AnASTERISK-26672: Crash when setting remote address on RTP instance
Reported by: Richard MudgettASTERISK-26617: res_rtp_asterisk: Can't bind on systems without IPv6
Reported by: Guido FalsiASTERISK-26566: res_rtp_asterisk: RTT miscalculation in RTCP
Reported by: Hector Royo ConcepcionASTERISK-26280: DNS lookups can block channel media paths
Reported by: Mark MichelsonASTERISK-26207: [patch] sRTP: Count a roll-over of the sequence number even on lost packets.
Reported by: Alexander TraudASTERISK-25659: res_rtp_asterisk: ECDH not negotiated causing DTLS failure occurred on RTP instance
Reported by: Edwin VandammeASTERISK-26140: res_rtp_asterisk: gcc 6 caught a self-comparison
Reported by: George JosephASTERISK-26129: res_rtp_asterisk: Memory leak of CERT bio in DTLS implementation
Reported by: Torrey SearleASTERISK-26130: [patch] WebRTC: Should use latest DTLS version.
Reported by: Alexander TraudASTERISK-26092: [Segfault] in res_rtp_asterisk.c:4268 after Remotely bridged channels
Reported by: Niklas LarssonASTERISK-25642: res_rtp_asterisk: SRTCP broken with DTLS - bad video is one of the consequences
Reported by: Stefan EngströmASTERISK-25645: res_rtp_asterisk: Lock inversion
Reported by: Steve DaviesASTERISK-24146: [patch]No audio on WebRtc caller side when answer waiting time is more than ~7sec
Reported by: Aleksei KulakovASTERISK-25451: Broken video - erased rtp marker bit
Reported by: Stefan EngströmASTERISK-25438: res_rtp_asterisk: ICE role message even when ICE is not enabled
Reported by: Joshua C. ColpASTERISK-25265: [patch]DTLS Failure when calling WebRTC-peer on Firefox 39 - add ECDH support and fallback to prime256v1
Reported by: Stefan EngströmASTERISK-25103: Roundup - investigate Asterisk DTLS crashes
Reported by: Rusty NewtonASTERISK-22805: res_rtp_asterisk: Crash when calling BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
Reported by: Dmitry BurilovASTERISK-24651: [patch] Fix race condition in DTLS
Reported by: Badalian VyacheslavASTERISK-24832: [patch]DTLS-crashes within openssl
Reported by: Stefan EngströmASTERISK-25127: DTLS crashes following "Unable to cancel schedule ID" in dtls_srtp_check_pending
Reported by: Dade BrandonASTERISK-25022: Memory leak setting up DTLS/SRTP calls
Reported by: Steve DaviesASTERISK-24791: Crash in ast_rtcp_write_report
Reported by: JoshEASTERISK-24337: Spammy DEBUG message needs to be at a higher level - 'Remote address is null, most likely RTP has been stopped'
Reported by: Rusty NewtonASTERISK-24604: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core
Reported by: Matt JordanASTERISK-24383: res_rtp_asterisk: Crash if no candidates received for component
Reported by: Kevin HarwellASTERISK-24326: res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted
Reported by: Joshua C. ColpASTERISK-23577: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL
Reported by: Jay JideliovASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Reported by: Matt Jordan

Category: Resources/res_rtp_multicast

ASTERISK-21399: RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
Reported by: Tzafrir CohenASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode

Category: Resources/res_security_log

ASTERISK-20744: [patch] Security event logging does not work over syslog
Reported by: Michael Keuter

Category: Resources/res_smdi

ASTERISK-19657: Coverity Report: Fix issues for error type CHAR_IO
Reported by: Matt JordanASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-24066: res_smdi: convert to astobj2
Reported by: Corey Farrell

Category: Resources/res_sorcery_memory_cache

ASTERISK-26731: res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
Reported by: Ustinov Artem

Category: Resources/res_srtp

ASTERISK-27905: [patch] res_srtp: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-27733: [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
Reported by: Alexander TraudASTERISK-27356: [patch] libsrtp-2.x.x + AES-GCM support
Reported by: Alexander TraudASTERISK-25294: srtp's crypto_get_random deprecated
Reported by: Tzafrir CohenASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
Reported by: Javier Riveros ASTERISK-24436: Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0
Reported by: Patrick LaimbockASTERISK-25642: res_rtp_asterisk: SRTCP broken with DTLS - bad video is one of the consequences
Reported by: Stefan EngströmASTERISK-24550: res_rtp_asterisk: Crash in ast_rtp_on_ice_complete during DTLS handshake
Reported by: Osaulenko Alexander

Category: Resources/res_stasis

ASTERISK-28333: StasisEnd event makes wrong timestamp value
Reported by: sungtae kimASTERISK-26094: stasis: Playing MOH to bridge with ARI does not work
Reported by: CameronASTERISK-27656: CDR: Leaking channel snapshots allocated by stasis_channel.c
Reported by: Kristijan VrbanASTERISK-27059: res_stasis: Stolen channel references are leaking
Reported by: George JosephASTERISK-26047: ARI allows certain commands to run on down channels.
Reported by: Mark MichelsonASTERISK-25947: Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object.
Reported by: Richard MudgettASTERISK-24649: Pushing of channel into bridge fails; Stasis fails to get app name
Reported by: John BigelowASTERISK-24782: StasisEnd event not present for channel that was swapped out for another after completing attended transfer
Reported by: John BigelowASTERISK-25882: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2)
Reported by: Richard MudgettASTERISK-25709: ARI: Crash can occur due to race condition when attempting to operate on a hung up channel
Reported by: Mark MichelsonASTERISK-25181: ARI: Channels added to Stasis application during WebSocket creation don't receive a StasisStart event
Reported by: Matt JordanASTERISK-24755: Asterisk sends unexpected early BYE to transferrer during attended transfer when using a Stasis bridge
Reported by: John BigelowASTERISK-24701: Stasis: Write timeout on WebSocket fails to fully disconnect underlying socket, leading to events being dropped with no additional information
Reported by: Matt JordanASTERISK-24637: Channel re-enters Stasis() when it should not
Reported by: John BigelowASTERISK-24537: Stasis: StasisStart/StasisEnd events are not reliably transmitted during transfers
Reported by: Matt Jordan

Category: Resources/res_stasis_device_state

ASTERISK-27130: Applications ARI: Unsubscribe action for deviceStates does not remove old subscriptions properly
Reported by: Sergej KasumovicASTERISK-26770: res_stasis_device_state: Duplicate subscriptions when multiple received at same time
Reported by: Joshua C. Colp

Category: Resources/res_stasis_playback

ASTERISK-26083: ARI: Announcer channels staying around after playback to a bridge is finished
Reported by: Per Jensen

Category: Resources/res_stasis_snoop

ASTERISK-27128: [patch]res_stasis_snoop: When recording a snoop channel (using ARI) where no media is being received, no recording happens when theres no media
Reported by: Dan JenkinsASTERISK-26973: bridge: Crash when freeing frame and snooping
Reported by: Michel R. VaillancourtASTERISK-24938: ARI Snoop Channel results in excessive escalating CPU usage
Reported by: George Ladoff

Category: Resources/res_statsd

ASTERISK-27389: Optional API modules should not allow unload.
Reported by: Corey FarrellASTERISK-25595: Unescaped : in messge sent to statsd
Reported by: Niklas Larsson

Category: Resources/res_stun_monitor

ASTERISK-21856: STUN never works when asterisk started without internet access
Reported by: Jeremy Kister

Category: Resources/res_timing_kqueue

ASTERISK-19277: [patch]endlessly repeating error: "poll failed: Bad file descriptor"
Reported by: Barry ChernASTERISK-24857: [patch] "timing test", pjsip incoming/outgoing calls, voicemail prompts and recordings all fail when using the kqueue timer source on FreeBSD 10.x
Reported by: Justin T. Gibbs

Category: Resources/res_timing_pthread

ASTERISK-24768: res_timing_pthread: file descriptor leak
Reported by: Matthias Urlichs

Category: Resources/res_timing_timerfd

ASTERISK-19277: [patch]endlessly repeating error: "poll failed: Bad file descriptor"
Reported by: Barry Chern

Category: Resources/res_xmpp

ASTERISK-27346: res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
Reported by: Ronald RaikesASTERISK-27207: XMPP OAuth not working due to inverted logic
Reported by: Michael KuronASTERISK-21009: xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
Reported by: Marcello CeschiaASTERISK-24712: xmpp: starttls problem causes connection spew
Reported by: Matthias UrlichsASTERISK-23510: JABBER_STATUS fails with improper code 7 for unavailable clients
Reported by: Anthony CritelliASTERISK-21855: Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
Reported by: Jeremy KisterASTERISK-25622: WARNING for "JABBER: socket read error" should be more specific
Reported by: Sean DarcyASTERISK-24425: [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
Reported by: abelbeckASTERISK-26164: XMPP no longer triggers NOTIFY to device via chan_pjsip
Reported by: Ross BeerASTERISK-25735: [patch] res_xmpp: Does not connect in component mode
Reported by: Karsten WemheuerASTERISK-24780: [patch] - Buddies are always auto-registered when processing the roster
Reported by: Simon Arlott

Category: Sounds

ASTERISK-16172: Problems with siren14 codec; problems with siren7 sound files.
Reported by: Steve MurphyASTERISK-25810: say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
Reported by: Nicolas RiendeauASTERISK-27142: sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
Reported by: Corey FarrellASTERISK-26807: sounds: New 3-D Binaural audio features require new sound prompts
Reported by: Rusty NewtonASTERISK-25816: French conf-adminmenu, conf-usermenu prompts differ in content from the English files
Reported by: Benoit DuvergerASTERISK-26274: Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
Reported by: Rusty Newton

Category: Tests/General

ASTERISK-28251: CI: Fix CI so it reverifies commit message changes
Reported by: George JosephASTERISK-28070: testsuite: Sniffer assumes pjmedia will use ports below 10000
Reported by: Joshua C. ColpASTERISK-27914: [patch] tests/test_utils: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-25960: The config_hook unit test causes Asterisk to crash if run a second time
Reported by: George JosephASTERISK-26739: voicemail API test: confuses expected and actual values
Reported by: Tzafrir CohenASTERISK-26740: voicemail API test: uses varlibdir instead of datadir for a sound file
Reported by: Tzafrir CohenASTERISK-26647: Support older DNS style for OpenBSD
Reported by: snuffyASTERISK-26211: Unit tests: AST_TEST_DEFINE should be used in conditional code.
Reported by: Corey FarrellASTERISK-26139: test_res_pjsip_scheduler: Compile failure if pjproject isn't installed in a system location
Reported by: George JosephASTERISK-25959: http_media_cache/retrieve_cache_control_directives: Sporadic failure
Reported by: Joshua C. ColpASTERISK-25685: infrastructure: Run alembic in Jenkins build script
Reported by: Joshua C. ColpASTERISK-25611: core: threadpool thread_timeout_thrash unit test sporadically failing
Reported by: Joshua C. ColpASTERISK-25053: Unit test category /main/presence missing trailing slash.
Reported by: Corey FarrellASTERISK-22367: Rework CEL unit test verification step
Reported by: Kinsey MooreASTERISK-24413: parking/parking_tests: Crash due to assertion in unit tests when MoH is started on channel in holding bridge
Reported by: Matt Jordan

Category: Tests/testsuite

ASTERISK-25961: tests/channels/SIP/sip_tls_call: Sporadic crash when running test
Reported by: Joshua C. ColpASTERISK-25582: Testsuite: Reactor timeout error in tests/fax/pjsip/directmedia_reinvite_t38
Reported by: Matt JordanASTERISK-25165: Testsuite - Sorcery memory cache leaks
Reported by: Corey FarrellASTERISK-25318: tests/rest_api/applications/subscribe-endpoint/nominal/resource: Sporadically failing
Reported by: Joshua C. ColpASTERISK-25292: Testuite: tests/apps/bridge/bridge_wait/bridge_wait_e_options fails
Reported by: Kevin HarwellASTERISK-25172: Crash in channels/sip/sip blind transfer/caller_refer_only test in ast_format_cap_append_from_cap during ast_request
Reported by: Matt JordanASTERISK-24212: testsuite: Sporadic crash due to assert on stopping RTP engine
Reported by: Matt JordanASTERISK-24215: testsuite: ARI Live Dangerously test fails due to wrong response code from Asterisk
Reported by: Matt Jordan

Category: Third-Party/pjproject

ASTERISK-28182: chan_pjsip: When connected_line_method is set to invite, asterisk is not trying UPDATE
Reported by: nappsoftASTERISK-27966: pjsip: Race condition in 183 re transmission can result in a deadlock
Reported by: Torrey SearleASTERISK-27880: [patch] pjproject_bundled: Repair ./configure --with-ssl=PATH.
Reported by: Alexander TraudASTERISK-27408: Identify causes and fix pjsip/resolver/srv/failover/in_dialog/transport_tcp
Reported by: Corey FarrellASTERISK-27097: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure
Reported by: George JosephASTERISK-26905: pjproject_bundled: Merge 3 upstream deadlock patches into bundled
Reported by: Ross BeerASTERISK-26872: Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
Reported by: Matt JordanASTERISK-26653: pjproject_bundled doesn't verify already downloaded tarballs
Reported by: George JosephASTERISK-26510: pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
Reported by: George JosephASTERISK-26477: pjproject: SEGV during SSL operations
Reported by: George JosephASTERISK-26279: pjproject-bundled: Fails to compile on Debian 6
Reported by: George Joseph

Category: Utilities/General

ASTERISK-13271: menuselect sets defaults too late
Reported by: John NemethASTERISK-23508: Memory Corruption in __ast_string_field_ptr_build_va
Reported by: Arnd Schmitter

Category: Utilities/aelparse

ASTERISK-27809: [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally.
Reported by: Alexander Traud

Category: Utilities/astcanary

ASTERISK-26352: Astcanary dies when doing "core restart"
Reported by: Walter DoekesASTERISK-19867: asterisk fails to lower its priority when astcanary dies
Reported by: Xavier Hienne

Category: Utilities/conf2ael

ASTERISK-27809: [patch] utils/pval: Add -lBlocksRuntime for compiler clang conditionally.
Reported by: Alexander Traud

Category: pjproject/pjsip

ASTERISK-28049: res_pjproject build failure
Reported by: Jaco KroonASTERISK-27997: pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
Reported by: Alexander TraudASTERISK-27961: res_pjsip: Spurious ERROR logging when printing headers in sip_msg
Reported by: Nick FrenchASTERISK-27584: Internal pjproject build doesn't disable bcg729
Reported by: Stuart HendersonASTERISK-24598: When running ./contrib/scripts/install_prereq install-unpackaged pjproject is installed in wrong place
Reported by: PowerPBXASTERISK-27391: Regression: Deadlock between AOR named lock and pjproject grp lock
Reported by: shaurya jainASTERISK-27001: res_pjsip: TLS connection not stable
Reported by: Ian GilmourASTERISK-27127: configs: Erroneous load directive in sample configuration results in "Error loading module 'res_pjsip_multihomed.so'"
Reported by: HZMI8gkCvPpom0tMASTERISK-27036: res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
Reported by: Maxim VasilevASTERISK-26939: Out of bound memory access in PJSIP multipart parser crashes Asterisk
Reported by: Sandro GauciASTERISK-26938: Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
Reported by: Sandro GauciASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i)
Reported by: Matthias BinderASTERISK-26930: pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
Reported by: abelbeckASTERISK-26814: pjproject_bundled build fails to download pjproject source when using cURL
Reported by: Gergely DömsödiASTERISK-26669: PJSIP Segfault 13.13.1 (Bundled PJSIP)
Reported by: Nic ColledgeASTERISK-26802: [patch] Integrity Check Of PJSIP Download Fails
Reported by: Michael L. YoungASTERISK-26696: pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
Reported by: Zach RASTERISK-26655: [patch]pjsip: Transfers Broken with Compact Headers Enabled
Reported by: JoshEASTERISK-26490: res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
Reported by: Juris BreicisASTERISK-26344: Asterisk 13.11.0 + PJSIP crash
Reported by: Ian GilmourASTERISK-26477: pjproject: SEGV during SSL operations
Reported by: George JosephASTERISK-26349: 13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
Reported by: Dmitry MelekhovASTERISK-26199: PJSIP: tx_data_destroy called twice
Reported by: Scott GriepentrogASTERISK-26089: Invalid security events during boot using PJSIP Realtime
Reported by: Scott GriepentrogASTERISK-25993: pjproject: Allow bundling to not require everything it does
Reported by: Joshua C. ColpASTERISK-25968: pjproject_bundled: Configure and make need to be re-tested
Reported by: George JosephASTERISK-25970: Segfault in pjsip_url_compare
Reported by: Dmitriy SerovASTERISK-25910: pjproject: Via headers are not parsed when "received" contains an IPv6 address
Reported by: George JosephASTERISK-25337: Crash on PJSIP_HEADER Add P-Asserted-Identity when calling from Gosub
Reported by: Jacques PeacockASTERISK-25615: res_pjsip: Setting transport async_operations > 1 causes segfault on tls transports
Reported by: George JosephASTERISK-25513: Crash: malloc failed with high load of subscriptions.
Reported by: John BigelowASTERISK-24602: Unable to call WebRTC client via wss on chan_pjsip
Reported by: Oleg KozlovASTERISK-24963: ASAN: heap-use-after-free with PJSIP and WSS
Reported by: Badalian VyacheslavASTERISK-25018: pjsip show endpoints crashes asterisk when qualified aors present
Reported by: Ivan PoddubnyASTERISK-24807: Missing mandatory field Max-Forwards
Reported by: AnatoliASTERISK-24471: Crash - assert_fail in libc in pjmedia_sdp_neg_negotiate from /usr/local/lib/libpjmedia.so.2
Reported by: yaron nahumASTERISK-24336: PJSIP timer_min_se value under 90 causes crash
Reported by: Leon Rowland

Improvement

Category: Addons/General

ASTERISK-27846: ast_coredumper: Fix OUTPUT directory
Reported by: Ted G

Category: Addons/chan_ooh323

ASTERISK-25129: wrong automatic ras address assignment if multihomed
Reported by: Dmitry MelekhovASTERISK-27793: cppcheck identifies redundant "if"
Reported by: Ilya Shipitsin

Category: Applications/app_authenticate

ASTERISK-23953: Testsuite: Off-nominal Authenticate test
Reported by: Matt Jordan

Category: Applications/app_confbridge

ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. ColpASTERISK-27877: app_confbridge: Add talking indicator for ConfBridgeList AMI response
Reported by: William McCallASTERISK-27651: app_confbridge: Add Muted to ConfbridgeJoin and channel snapshot headers to ConfbridgeList AMI events
Reported by: Richard MudgettASTERISK-27647: app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking.
Reported by: Richard MudgettASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis
Reported by: Dennis GuseASTERISK-26289: Announcer channels in ConfBridges cause inefficiencies
Reported by: Mark MichelsonASTERISK-24351: [patch] Allow passing options and command to MixMonitor when recording in ConfBridge
Reported by: Gareth Palmer

Category: Applications/app_controlplayback

ASTERISK-26562: app_controlplayback: Transmit Silence on ControlPlayback pause
Reported by: Mikheili Dautashvili

Category: Applications/app_dial

ASTERISK-26059: [patch]core: New channel variable FORWARDERNAME
Reported by: Alexei Gradinari

Category: Applications/app_followme

ASTERISK-24372: [patch] Add config option to play a prompt to the "winner" in app_followme
Reported by: Graham MainwaringASTERISK-26064: followme: allow disabling callee prompt
Reported by: Tzafrir Cohen

Category: Applications/app_meetme

ASTERISK-27873: documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event
Reported by: Alessandro Polidori

Category: Applications/app_mp3

ASTERISK-27752: Ten seconds of silence after mp3 playback
Reported by: Sam Wierema

Category: Applications/app_originate

ASTERISK-22992: [patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
Reported by: Anthony Messina

Category: Applications/app_queue

ASTERISK-27483: Allow wrapuptime to be set for each queue member
Reported by: Rodrigo Ramirez NorambuenaASTERISK-28055: app_queue: Per-member wrapup time missing from AddQueueMember application
Reported by: Niksa BaldunASTERISK-27912: [PATCH] Add predial handler to app_queue
Reported by: Kristian HøghASTERISK-27092: [patch] app_queue: Add Priority to AMI QueueStatus
Reported by: Niklas LarssonASTERISK-26559: app_queue: New service level calculation
Reported by: Sebastian GutierrezASTERISK-26558: app_queue: add variable to know if the call is not answered after a queue
Reported by: Sebastian GutierrezASTERISK-25581: [patch]Add value reason a pause on CLI
Reported by: Rodrigo Ramirez NorambuenaASTERISK-24365: [Patch] Dialplan function to get first/head caller channel on queue
Reported by: Kristian Høgh

Category: Applications/app_record

ASTERISK-24530: [patch] app_record stripping 1/4 second from recordings
Reported by: Ben Smithurst

Category: Applications/app_stasis

ASTERISK-24802: stasis: set a channel variable on websocket disconnect error
Reported by: Kevin Harwell

Category: Applications/app_voicemail

ASTERISK-27456: app_voicemail: Add new object for VoicemailUserEntry
Reported by: sungtae kimASTERISK-24790: Reduce spurious noise in logs from voicemail - Couldn't find mailbox %s in context
Reported by: Graham Barnett

Category: Applications/app_voicemail/IMAP

ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect
Reported by: Alexei GradinariASTERISK-26229: [patch] app_voicemail: Add taskprocessor alert level options.
Reported by: Alexei Gradinari

Category: Applications/app_voicemail/NewFeature

ASTERISK-27470: Add new object for VoicemailUserEntry
Reported by: sungtae kimASTERISK-24045: [patch]Voicemail to email at multiple email addresses
Reported by: Jacob Barber

Category: Bridges/bridge_builtin_features

ASTERISK-28279: Added creation timestamp for bridge
Reported by: sungtae kim

Category: Bridges/bridge_softmix

ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. ColpASTERISK-28196: bridge_softmix: Does not support WebRTC source with multi video tracks.
Reported by: Xiemin ChenASTERISK-27647: app_confbridge/bridge_softmix: When channel muted report talking stopped if was talking.
Reported by: Richard MudgettASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis
Reported by: Dennis Guse

Category: CDR/General

ASTERISK-24297: cdr.c: Minor code optimizations.
Reported by: Richard MudgettASTERISK-27335: CDR performance needs improvement.
Reported by: Richard MudgettASTERISK-24279: Documentation: Clarify the behaviour of the CDR property 'unanswered'
Reported by: Matt Jordan

Category: CDR/cdr_adaptive_odbc

ASTERISK-25109: [patch] CEL and CDR - Assigned separator for column name and values.
Reported by: Rodrigo Ramirez NorambuenaASTERISK-24980: cdr_adaptive_odbc: refactor lines to concatenate of columns name
Reported by: Rodrigo Ramirez Norambuena

Category: CDR/cdr_manager

ASTERISK-24671: Missing docs for the CDR AMI Event
Reported by: Dan Jenkins

Category: CDR/cdr_pgsql

ASTERISK-25109: [patch] CEL and CDR - Assigned separator for column name and values.
Reported by: Rodrigo Ramirez Norambuena

Category: CDR/cdr_radius

ASTERISK-26540: cdr_radius: use radcli instead of freeradius-client
Reported by: Tzafrir Cohen

Category: CEL/cel_odbc

ASTERISK-25109: [patch] CEL and CDR - Assigned separator for column name and values.
Reported by: Rodrigo Ramirez NorambuenaASTERISK-24283: [patch]Microseconds precision in the eventtime column in the cel_odbc module
Reported by: Etienne Lessard

Category: CEL/cel_pgsql

ASTERISK-24965: cel_pgsql - log_error string references CDR instead of CEL
Reported by: Rodrigo Ramirez Norambuena

Category: Channels/chan_dahdi

ASTERISK-28317: Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function
Reported by: Cirillo Ferreira

Category: Channels/chan_iax2

ASTERISK-24939: [patch]IAX make calltoken expiration time configurable
Reported by: Y Ateya

Category: Channels/chan_motif

ASTERISK-27169: Google OAuth 2.0 support for XMPP / Motif
Reported by: Andrey

Category: Channels/chan_pjsip

ASTERISK-28292: Changed to show all channel stats including wrong media
Reported by: sungtae kimASTERISK-28144: [patch] New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI
Reported by: Alexei GradinariASTERISK-27697: Enable in-dialog NOTIFY on chan_pjsip channels
Reported by: Nathan BruningASTERISK-27220: Enable CHANNEL function to get from and to tag from SIP Headers
Reported by: Andre NazarioASTERISK-27085: [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
Reported by: Torrey SearleASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey SearleASTERISK-22131: Update the make dependencies script to pull, build, and install the correct pjproject
Reported by: Matt JordanASTERISK-25471: [patch]Add subscribe_context to res_pjsip
Reported by: JoshEASTERISK-25835: Authentication using 'Username' field from Digest
Reported by: Ross BeerASTERISK-24706: [patch]add auto-dtmf mode for pjsip
Reported by: yaron nahumASTERISK-24862: [patch] Support in-dialog OPTIONS
Reported by: yaron nahum

Category: Channels/chan_sip/General

ASTERISK-27278: [patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
Reported by: David J. PrykeASTERISK-26846: chan_sip: Add rtcp-mux support
Reported by: Sean BrightASTERISK-26176: chan_sip: Add AccountCode to AMI PeerEntry
Reported by: Sebastian Gutierrez

Category: Channels/chan_sip/Interoperability

ASTERISK-27461: 3PCC patch for AMI "SIPnotify"
Reported by: Yasuhiko Kamata

Category: Channels/chan_sip/NewFeature

ASTERISK-25578: [patch] SIP/SDP: No rtpmap for static RTP payload IDs
Reported by: Alexander Traud

Category: Channels/chan_sip/Registration

ASTERISK-20527: AuthID cannot be set for registrations when callbackexten is used
Reported by: Timo Teräs

Category: Channels/chan_sip/Subscriptions

ASTERISK-25558: [patch]chan_sip option 'notifyringing' doc fix and addition of 'notifyringingprio'
Reported by: Ward van Wanrooij

Category: Channels/chan_sip/TCP-TLS

ASTERISK-24815: [patch] Enable TLS Dual-Certificates (ECC+RSA)
Reported by: Alexander TraudASTERISK-25043: [patch] Avoiding ERR_remove_state in OpenSSL
Reported by: Alexander Traud

Category: Channels/chan_sip/WebSocket

ASTERISK-24128: [Patch] Adding default dtls settings
Reported by: Michael K.

Category: Codecs/General

ASTERISK-26217: [patch] Codec 2 Mode 2400
Reported by: Alexander TraudASTERISK-26218: [patch] iLBC 20
Reported by: Alexander Traud

Category: Codecs/codec_lpc10

ASTERISK-23556: Compilation warning for invert.c (array subscript is above array bounds)
Reported by: Marcello Ceschia

Category: Codecs/codec_opus

ASTERISK-26538: codec_opus: Add sample to configs/samples/codecs.conf.sample
Reported by: Kevin Harwell

Category: Contrib/General

ASTERISK-28136: Allow the sip_to_pjsip script to be used in a pipe
Reported by: Pascal Cadotte MichaudASTERISK-27770: [patch] install_prereq: Add Slackware (somehow).
Reported by: Alexander TraudASTERISK-27769: [patch] install_prereq: Add Gentoo Linux.
Reported by: Alexander TraudASTERISK-27738: [patch] install_prereq: Add Arch Linux.
Reported by: Alexander TraudASTERISK-27736: [patch] install_prereq: Add SUSE.
Reported by: Alexander TraudASTERISK-27729: [patch] install_prereq: Add NetBSD.
Reported by: Alexander TraudASTERISK-27348: [patch]contrib/scripts: add a way to migrate from chan_sip to chan_pjsip realtime
Reported by: Torrey SearleASTERISK-27380: ast_coredumper: allow pointing out the asterisk binary explicitly
Reported by: Tzafrir CohenASTERISK-27255: alembic: Add support for Microsoft SQL server
Reported by: Florian FloimairASTERISK-25495: [patch] Prevent old-update packages on repository Debian systems
Reported by: Rodrigo Ramirez Norambuena

Category: Core/AstMM

ASTERISK-24974: Astobj2: Allow reference debugging to be enabled/disabled by config.
Reported by: Corey Farrell

Category: Core/Bridging

ASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis
Reported by: Dennis GuseASTERISK-26059: [patch]core: New channel variable FORWARDERNAME
Reported by: Alexei Gradinari

Category: Core/Bridging/bridge_basic

ASTERISK-27449: [PATCH] When failing to acquire target during attended transfer, display wanted extension
Reported by: Niklas Larsson

Category: Core/BuildSystem

ASTERISK-27929: [patch] BuildSystem: Enable autotools in Solaris 11.
Reported by: Alexander TraudASTERISK-27820: [patch] Add DragonFly BSD.
Reported by: Alexander TraudASTERISK-27728: [patch] BuildSystem: Add NetBSD.
Reported by: Alexander TraudASTERISK-27683: [patch] BuildSystem: Allow newer autotools on OpenBSD.
Reported by: Alexander TraudASTERISK-23556: Compilation warning for invert.c (array subscript is above array bounds)
Reported by: Marcello CeschiaASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL
Reported by: Guido FalsiASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis
Reported by: Dennis GuseASTERISK-26220: Add support for noreturn function attributes.
Reported by: Corey FarrellASTERISK-24718: [patch]Add inital support of "sanitize" to configure
Reported by: Badalian VyacheslavASTERISK-24960: Build System: Create MOD_ADD_SOURCE macro for module Makefiles
Reported by: Corey FarrellASTERISK-24133: [patch]Please support Clang; Allow no-exec stacks
Reported by: Jeffrey Walton

Category: Core/Channels

ASTERISK-26419: audiohooks: Remove redundant codec translations when using audiohooks
Reported by: Michael WaltonASTERISK-26059: [patch]core: New channel variable FORWARDERNAME
Reported by: Alexei Gradinari

Category: Core/General

ASTERISK-27867: [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated.
Reported by: Alexander TraudASTERISK-26419: audiohooks: Remove redundant codec translations when using audiohooks
Reported by: Michael WaltonASTERISK-26398: core: Remove ABI differences of LOW_MEMORY
Reported by: Corey FarrellASTERISK-25627: Easily Preventable Compile Warning
Reported by: Diederik de GrootASTERISK-25518: taskprocessor: Add high water mark
Reported by: Jonathan RoseASTERISK-25310: [patch]on FreeBSD also pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED
Reported by: Guido FalsiASTERISK-25256: [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable.
Reported by: Richard MudgettASTERISK-25063: [patch]add X.509 subject alternative name support to Asterisk TLS support
Reported by: Maciej SzmigieroASTERISK-25045: vector: Add new capabilities and unit tests
Reported by: George JosephASTERISK-25049: CLI: Enable automatic references to modules
Reported by: Corey FarrellASTERISK-25056: Modules: Make ast_module_info->self available to auxiliary sources.
Reported by: Corey FarrellASTERISK-24917: [patch] clang compilation warnings
Reported by: Diederik de GrootASTERISK-25051: Remove unneeded uses of optional_api providers.
Reported by: Corey FarrellASTERISK-24813: asterisk.c: #if statement in listener() confuses code folding editors
Reported by: Corey Farrell

Category: Core/HTTP

ASTERISK-27173: Support for GMIME 3.0
Reported by: Tzafrir CohenASTERISK-24316: For httpd server, need option to define server name for security purposes
Reported by: Andrew Nagy

Category: Core/ManagerInterface

ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot output
Reported by: Matt Jordan

Category: Core/ManagerInterface/NewFeature

ASTERISK-24730: [patch] Add blank line between headers and output for Command action response
Reported by: Gareth Palmer

Category: Core/PBX

ASTERISK-26658: Add ability for dialplan show to display filenames/line numbers of registered extensions
Reported by: Jonathan R. RoseASTERISK-25040: pbx: Improve performance of reloads by making hint destruction more performant
Reported by: Matt JordanASTERISK-24038: device state: Report ONHOLD device state if channel driver defers device state calculation to core
Reported by: Matt Jordan

Category: Core/Portability

ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
Reported by: Guido FalsiASTERISK-24517: TLS support for Solaris, Ming and non-glibc Linux systems
Reported by: Timo Teräs

Category: Core/Sorcery

ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard MudgettASTERISK-25067: Sorcery Caching: Implement a new caching module
Reported by: Matt JordanASTERISK-25044: sorcery: Add ability to insert a new wizard into an object type's list
Reported by: George Joseph

Category: Core/Stasis

ASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard Mudgett

Category: Documentation

ASTERISK-27993: pjsip_wizard example gives wrong info about unsupported SRV records
Reported by: Jonathan HarrisASTERISK-27873: documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event
Reported by: Alessandro PolidoriASTERISK-24892: Super Awesome Company sound prompts
Reported by: Rusty NewtonASTERISK-24671: Missing docs for the CDR AMI Event
Reported by: Dan JenkinsASTERISK-23512: Inaccurate comment in manager.conf.sample
Reported by: Richard MillerASTERISK-24171: [patch] Provide a manpage for the aelparse utility
Reported by: Jeremy Lainé

Category: Features

ASTERISK-25405: [patch] CLI: core show fd: add timestamp
Reported by: Alexander TraudASTERISK-24678: [PATCH] Added atxfer* settings to features.conf.sample
Reported by: Niklas Larsson

Category: Features/Parking

ASTERISK-22825: Dialplan Function for Checking Parking Lot Slot
Reported by: JoshE

Category: Formats/NewFeature

ASTERISK-26292: app_confbridge: 3D-Conferencing via Binaural Synthesis
Reported by: Dennis Guse

Category: Formats/format_g726

ASTERISK-28246: Support skipping on the g726 format
Reported by: Eyal Hasson

Category: Functions/func_odbc

ASTERISK-26010: [patch]func_odbc: single database connection should be optional
Reported by: Alexei Gradinari

Category: General

ASTERISK-28046: Remove stale nonoptreq references
Reported by: Walter DoekesASTERISK-27014: configurable busy_timeout in sqlite backends
Reported by: Marek CervenkaASTERISK-25846: Gracefully deal with Absent Stasis Apps
Reported by: Andrew NagyASTERISK-25767: [patch] Add check to configure for sanitizes
Reported by: Badalian VyacheslavASTERISK-25376: Scripts: check file versions for Asterisk and dependencies
Reported by: Scott GriepentrogASTERISK-24745: [patch]Add no_answer to ARI hangup causes
Reported by: Ben Merrills

Category: PBX/NewFeature

ASTERISK-27661: Add new AMI Event for Load, Unload
Reported by: sungtae kim

Category: PBX/pbx_config

ASTERISK-27084: Reduce verbosity while loading PBX extensions.
Reported by: Ludovic Gasc (Eyepea)ASTERISK-26658: Add ability for dialplan show to display filenames/line numbers of registered extensions
Reported by: Jonathan R. Rose

Category: PBX/pbx_dundi

ASTERISK-27164: [patch] Add IPv6 Support for DUNDi
Reported by: Adam Secombe

Category: PBX/pbx_loopback

ASTERISK-24577: Speed up loopback switches by avoiding unneeded lookups
Reported by: Birger "WIMPy" Harzenetter

Category: PBX/pbx_spool

ASTERISK-26568: pbx_spool: OUTGOING_RETRY variable
Reported by: Roman Shubovich

Category: Resources/res_agi

ASTERISK-26124: res_agi: Set audio format for EAGI audio stream
Reported by: John Fawcett

Category: Resources/res_ari

ASTERISK-28326: ari: Added timestamp for some ari events.
Reported by: sungtae kimASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE command
Reported by: Sebastian DammASTERISK-26488: ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
Reported by: Matt JordanASTERISK-24802: stasis: set a channel variable on websocket disconnect error
Reported by: Kevin HarwellASTERISK-24553: ARI/AMI: Include language in standard channel snapshot output
Reported by: Matt JordanASTERISK-24552: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
Reported by: Matt Jordan

Category: Resources/res_ari_applications

ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan

Category: Resources/res_ari_bridges

ASTERISK-24870: ARI: Subscriptions to bridges generally not super useful
Reported by: Matt Jordan

Category: Resources/res_ari_channels

ASTERISK-28198: res_ari: Add new hangup causes for ARI Channel DELETE command
Reported by: Sebastian DammASTERISK-26321: ARI : Add reason answered_elsewhere to channel hangup
Reported by: Jean Aunis - PrescomASTERISK-24412: [patch]Incomplete channel originate/continue handling with ARI
Reported by: Nir Simionovich (GreenfieldTech - Israel)ASTERISK-24552: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes
Reported by: Matt Jordan

Category: Resources/res_calendar/NewFeature

ASTERISK-26422: [patch] Force calendars to do new fetch after module reload
Reported by: Ludovic Gasc (Eyepea)

Category: Resources/res_calendar_caldav

ASTERISK-26624: res_calendar_caldav: Add support for gmail
Reported by: Eduardo Scudeller Libardi

Category: Resources/res_config_pgsql

ASTERISK-25132: escaping manually
Reported by: Rodrigo Ramirez Norambuena

Category: Resources/res_crypto

ASTERISK-27906: [patch] res_crypto: Allow OpenSSL configured with no-deprecated.
Reported by: Alexander Traud

Category: Resources/res_fax

ASTERISK-25980: [patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
Reported by: Alexei Gradinari

Category: Resources/res_format_attr_opus

ASTERISK-26409: codec_opus: Update Asterisk to support the translation codec.
Reported by: Kevin Harwell

Category: Resources/res_hep

ASTERISK-27796: res_hep: Allow create_address to resolve a provided hostname
Reported by: Sebastian GutierrezASTERISK-26159: res_hep: enabled by default and information sent to default address
Reported by: Ross Beer

Category: Resources/res_hep_rtcp

ASTERISK-26427: res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
Reported by: Nir Simionovich (GreenfieldTech - Israel)

Category: Resources/res_musiconhold

ASTERISK-25444: [patch]Music On Hold Warning misleading
Reported by: Conrad de Wet

Category: Resources/res_pjsip

ASTERISK-27537: res_pjsip: Add new AMI Action for PJSIPShowAors
Reported by: sungtae kimASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode
Reported by: Torrey SearleASTERISK-26088: Investigate heavy memory utilization by res_pjsip_pubsub
Reported by: Richard MudgettASTERISK-23828: pjsip - Need a command to list active SIP subscriptions
Reported by: Rusty NewtonASTERISK-26011: [patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts
Reported by: Alexei GradinariASTERISK-2