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asterisk/asterisk-19.0.0-rc1-summary.txt
Asterisk Development Team 6022157d76 Update for 19.0.0-rc1
2021-10-13 05:44:39 -05:00

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Release Summary
asterisk-19.0.0-rc1
Date: 2021-10-13
<asteriskteam@digium.com>
----------------------------------------------------------------------
Table of Contents
1. Summary
2. Contributors
3. Closed Issues
4. Open Issues
5. Other Changes
6. Diffstat
----------------------------------------------------------------------
Summary
[Back to Top]
This is the first release of a major new version of Asterisk. For a list
of new features that have been included with this release, please see the
CHANGES file inside the source package. Since this is a new major release,
users are encouraged to do extended testing before upgrading to this
version in a production environment.
The data in this summary reflects changes that have been made since the
previous release, asterisk-18.0.0.
----------------------------------------------------------------------
Contributors
[Back to Top]
This table lists the people who have submitted code, those that have
tested patches, as well as those that reported issues on the issue tracker
that were resolved in this release. For coders, the number is how many of
their patches (of any size) were committed into this release. For testers,
the number is the number of times their name was listed as assisting with
testing a patch. Finally, for reporters, the number is the number of
issues that they reported that were affected by commits that went into
this release.
Coders Testers Reporters
63 Sean Bright 2 Mark Petersen 41 N A
61 Joshua C. Colp 1 Joseph Nadiv 33 Joshua C. Colp
42 Naveen Albert 16 Alexander Traud
37 George Joseph 11 George Joseph
30 Alexander Traud 8 sungtae kim
17 Kevin Harwell 6 Sean Bright
16 Ben Ford 5 Jean Aunis - Prescom
14 Jaco Kroon 5 Boris P. Korzun
5 Torrey Searle 4 Michael Maier
5 Sungtae Kim 4 Ross Beer
5 Ivan Poddubnyi 4 Sebastian Damm
4 Boris P. Korzun 3 Dan Cropp
4 Jean Aunis 3 Matthias Hensler
3 Nick French 3 Andre Barbosa
3 Mark Murawski 3 Ivan Poddubny
3 Sebastien Duthil 3 Sébastien Duthil
3 Joseph Nadiv 3 Torrey Searle
3 Andre Barbosa 3 Dan Cropp
2 sungtae kim 2 under
2 Dan Cropp 2 Jaco Kroon
2 Bernd Zobl 2 Caesar
2 Alexei Gradinari 2 Luke Escude
2 Richard Mudgett 2 Robert Sutton
2 Holger Hans Peter Freyther 2 Alexander Traud
2 Igor Goncharovsky 2 Rusty Newton
2 laszlovl 2 Kevin Harwell
1 Sarah Autumn 2 Igor Goncharovsky
1 Nathan Bruning 2 Andrew Yager
1 Pirmin Walthert 2 Mark Petersen
1 Rijnhard Hessel 2 Gregory Massel
1 Stanislav 2 Mark Petersen
1 Matthew Kern 2 laszlovl
1 Walter Doekes 2 Brian J. Murrell
1 Asterisk Development Team 2 Nick French
1 Jasper van der Neut 2 Stefan Ruf
1 Dennis Buteyn 1 Michael Welk
1 Nico Kooijman 1 Walter Doekes
1 under 1 tootai
1 Guido Falsi 1 Juan Carlos Castro y Castro
1 Andrew Siplas 1 Jacek Konieczny
1 Mark Petersen 1 Julien
1 Kfir Itzhak 1 Vyrva Igor
1 Michael Neuhauser 1 Sta Retji
1 Salah Ahmed 1 Joseph Nadiv
1 Jeremy Lainé 1 Ramarajan
1 Carlos Oliva 1 Benjamin Keith Ford
1 Evandro César Arruda 1 dovid
1 Shloime Rosenblum 1 Marco Paland
1 Michal Hajek 1 Lucas Tardioli Silveira
1 Alexander Greiner-Baer 1 N GM
1 Nickolay Shmyrev 1 Jeremy Lainé
1 Dovid Bender 1 Roman Pertsev
1 cmaj 1 Igor Liferenko
1 Patrick Verzele 1 Francisco Correia
1 Jasper Hafkenscheid 1 Corey Farrell
1 Robert Cripps 1 Michael Neuhauser
1 Evgenios_Greek 1 Ivan Poddubny
1 Thomas Johnson
1 Thomas Frederiksen
1 Vitezslav Novy
1 Etienne Lessard
1 Andrea Sannucci
1 siggi
1 Asterisk to be misaligned.
1 Evandro César Arruda
1 Matthew Kern
1 Michal Hajek
1 Mikhail Ivanov
1 Sarah Autumn
1 周家建
1 Edvin Vidmar
1 Hendrik Wedhorn
1 Salah Ahmed
1 Guido Falsi
1 N A
1 Michael
1 Péter Juhász
1 David Cunningham
1 Dennis
1 Bernd Zobl
1 Nathan Bruning
1 Alex Hermann
1 Michael Munger
1 Vieri
1 Tomas Maldonado
1 Rijnhard Hessel
1 Chris
1 Stanislav Abramenkov
1 Miguel Sanz
1 Isaac McDonald
1 Ove Aursand
1 Alexander Zharov
1 cmaj
1 bbawkon
1 Hajek Michal
1 Carlos Oliva
1 Alexander Gonchiy
1 Benjamin M.
1 Walter Doekes
1 Alex Hermann
1 Francisco Correia
1 Schneur Rosenberg
1 Philip Young
1 Alexander Akimov
1 Misha Vodsedalek
1 Dalius Mockevicius
1 Dovid Bender
1 Joseph Ades
1 Jasper van der Neut
1 Michael Newton
1 Alexander Greiner-Baer
1 Mauri de Souza Meneguzzo
(3CPlus)
1 Gant Liu
1 Nickolay V. Shmyrev
1 Eric Smith
1 Flole Systems
1 Alexei Gradinari
1 Michael Maier
1 Boolah
1 Andrew Siplas
1 Shloime Rosenblum
1 Brian J. Murrell
1 Ernani José Camargo Azevedo
1 Jacek Konieczny
1 Lucas Tardioli Silveira
1 IAMJames_
1 Leandro Dardini
1 Michael Neuhauser
1 Sandro Gauci
1 Charlie Smurthwaite
1 Brian Paboojian
1 Mark Murawski
1 Jasper Hafkenscheid
1 Robert Cripps
1 Kfir Itzhak
----------------------------------------------------------------------
Closed Issues
[Back to Top]
This is a list of all issues from the issue tracker that were closed by
changes that went into this release.
Deprecation
Category: Addons/app_mysql
ASTERISK-29585: app_mysql: Remove deprecated module
Reported by: Joshua C. Colp
* [1961a1b83e] Joshua C. Colp -- app_mysql: Remove deprecated module.
Category: Addons/cdr_mysql
ASTERISK-29584: cdr_mysql: Remove deprecated module
Reported by: Joshua C. Colp
* [3e07b1ff62] Joshua C. Colp -- cdr_mysql: Remove deprecated module.
Category: Applications/app_dahdiras
ASTERISK-29591: app_dahdiras: Remove deprecated module
Reported by: Joshua C. Colp
* [f18107f191] Joshua C. Colp -- app_dahdiras: Remove deprecated module.
Category: Applications/app_fax
ASTERISK-29587: app_fax: Remove deprecated module
Reported by: Joshua C. Colp
* [41afcb9422] Joshua C. Colp -- app_fax: Remove deprecated module.
Category: Applications/app_ices
ASTERISK-29586: app_ices: Remove deprecated module
Reported by: Joshua C. Colp
* [83cad340fc] Joshua C. Colp -- app_ices: Remove deprecated module.
Category: Applications/app_image
ASTERISK-29589: app_image: Remove deprecated module
Reported by: Joshua C. Colp
* [7ee6fb0372] Joshua C. Colp -- app_image: Remove deprecated module.
Category: Applications/app_macro
ASTERISK-29558: app_macro: Deprecated in 16, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Applications/app_meetme
ASTERISK-29548: app_meetme: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Applications/app_nbscat
ASTERISK-29590: app_nbscat: Remove deprecated module
Reported by: Joshua C. Colp
* [b1e5b1874c] Joshua C. Colp -- app_nbscat: Remove deprecated module.
Category: Applications/app_osplookup
ASTERISK-29549: app_osploop: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Applications/app_url
ASTERISK-29588: app_url: Remove deprecated module
Reported by: Joshua C. Colp
* [0b3a149001] Joshua C. Colp -- app_url: Remove deprecated module.
Category: CDR/cdr_syslog
ASTERISK-29592: cdr_syslog: Remove deprecated module
Reported by: Joshua C. Colp
* [e4b6f24a1d] Joshua C. Colp -- cdr_syslog: Remove deprecated module.
Category: Channels/chan_alsa
ASTERISK-29601: moduleinfo: Add replacement module information
Reported by: N A
* [432fe9dc2a] Naveen Albert -- chan_alsa, chan_sip: Add replacement to
moduleinfo
ASTERISK-29550: chan_alsa: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Channels/chan_mgcp
ASTERISK-29551: chan_mgcp: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Channels/chan_misdn
ASTERISK-29596: chan_misdn: Remove deprecated module
Reported by: Joshua C. Colp
* [72a2140a50] Joshua C. Colp -- chan_misdn: Remove deprecated module.
Category: Channels/chan_nbs
ASTERISK-29595: chan_nbs: Remove deprecated module
Reported by: Joshua C. Colp
* [7b0d3d3550] Joshua C. Colp -- chan_nbs: Remove deprecated module.
Category: Channels/chan_oss
ASTERISK-29593: chan_oss: Remove deprecated module
Reported by: Joshua C. Colp
* [d0ad32c7cf] Joshua C. Colp -- chan_oss: Remove deprecated module.
Category: Channels/chan_phone
ASTERISK-29594: chan_phone: Remove deprecated module
Reported by: Joshua C. Colp
* [7361a52820] Joshua C. Colp -- chan_phone: Remove deprecated module.
Category: Channels/chan_sip/General
ASTERISK-29601: moduleinfo: Add replacement module information
Reported by: N A
* [432fe9dc2a] Naveen Albert -- chan_alsa, chan_sip: Add replacement to
moduleinfo
ASTERISK-29567: chan_sip: Deprecated in 17, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Channels/chan_skinny
ASTERISK-29552: chan_skinny: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Channels/chan_vpb
ASTERISK-29597: chan_vpb: Remove deprecated module
Reported by: Joshua C. Colp
* [9d5f55a5f3] Joshua C. Colp -- chan_vpb: Remove deprecated module.
Category: General
ASTERISK-29599: conf2ael: Remove deprecated application
Reported by: Joshua C. Colp
* [650cf0b444] Joshua C. Colp -- conf2ael: Remove deprecated
application.
Category: Resources/General
ASTERISK-29553: res_pktccops: Deprecated in 19, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Resources/res_config_sqlite
ASTERISK-29598: res_config_sqlite: Remove deprecated module
Reported by: Joshua C. Colp
* [368aa47962] Joshua C. Colp -- res_config_sqlite: Remove deprecated
module.
Category: Resources/res_monitor
ASTERISK-29602: res_monitor: Disable building by default.
Reported by: Joshua C. Colp
* [ecf699c325] Joshua C. Colp -- res_monitor: Disable building by
default.
ASTERISK-29572: res_monitor: Deprecated in 16, to be removed in 21
Reported by: Joshua C. Colp
* [141dc519b0] Joshua C. Colp -- policy: Deprecate modules and add
versions to others.
Category: Utilities/muted
ASTERISK-29600: muted: Remove deprecated application
Reported by: Joshua C. Colp
* [daca793ad4] Joshua C. Colp -- muted: Remove deprecated application.
Security
Category: Channels/chan_pjsip
ASTERISK-29415: Crash in PJSIP TLS transport
Reported by: Andrew Yager
* [151bdbc658] Kevin Harwell -- AST-2021-009 - pjproject-bundled: Avoid
crash during handshake for TLS
Category: Resources/res_pjsip_diversion
ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains
History-Info
Reported by: Torrey Searle
* [51e2187a14] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
Category: Resources/res_pjsip_session
ASTERISK-29381: chan_pjsip: Remote denial of service by an authenticated
user
Reported by: Ivan Poddubny
* [45af7e9984] Joshua C. Colp -- AST-2021-007 - res_pjsip_session: Don't
offer if no channel exists.
Category: Resources/res_pjsip_t38
ASTERISK-29305: ASTERISK-29203 / AST-2021-002 -- Another scenario is
causing a crash
Reported by: Gregory Massel
* [fd560ad9fa] Ben Ford -- AST-2021-006 - res_pjsip_t38.c: Check for
session_media on reinvite.
Category: Resources/res_srtp
ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
Reported by: Alexander Traud
* [389b8b0774] Alexander Traud -- rtp: Enable srtp replay protection
Category: pjproject/pjsip
ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes
memory corruption and crash
Reported by: Ivan Poddubny
* [7d15655f9d] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more
than one histinfo to Supported
ASTERISK-29057: pjsip: Crash on call rejection during high load
Reported by: Sandro Gauci
* [b82f880647] Kevin Harwell -- AST-2020-001 - res_pjsip: Return dialog
locked and referenced
New Feature
Category: Applications/NewFeature
ASTERISK-29496: Add SendMF application
Reported by: N A
* [203e73f5af] Naveen Albert -- app_mf: Add channel agnostic MF sender
ASTERISK-29454: New application to reload modules
Reported by: N A
* [244491f9b2] Naveen Albert -- app_reload: New Reload application
ASTERISK-29444: Add application to wait for condition
Reported by: N A
* [c01b4e0d4b] Naveen Albert -- app_waitforcond: New application
Category: Applications/app_confbridge
ASTERISK-29446: app_confbridge: New ConfKick application
Reported by: N A
* [35437879e5] Naveen Albert -- app_confbridge: New ConfKick()
application
ASTERISK-29440: app_confbridge: Allow ConfBridge answer to be suppressed
Reported by: N A
* [5f8cabc232] Naveen Albert -- app_confbridge: New option to prevent
answer supervision
Category: Applications/app_dial
ASTERISK-29442: app_dial: Expand A option to allow announcement playback
to caller
Reported by: N A
* [1e5a2cfe30] Naveen Albert -- app_dial: Expanded A option to add
caller announcement
Category: Applications/app_read
ASTERISK-18454: Option for Read to be able to accept #
Reported by: Sta Retji
* [0e4a1c5079] Naveen Albert -- app_read: Allow reading # as a digit
Category: Channels/chan_pjsip
ASTERISK-27477: Chan_pjsip does not support unauthenticated OPTIONS ping
Reported by: Ross Beer
* [4a843e00ef] Sean Bright -- res_pjsip.c: OPTIONS processing can now
optionally skip authentication
Category: Core/General
ASTERISK-11: AGI channel_status failure
Reported by: bbawkon
* [ff8ca2c9f1] under -- codec_builtin.c: G729 audio gets corrupted by
Asterisk due to smoother
Category: Functions/NewFeature
ASTERISK-29531: Add SAYFILES function
Reported by: N A
* [0b8ae58e67] Naveen Albert -- func_sayfiles: Retrieve say file names
ASTERISK-29542: Add audio scrambler
Reported by: N A
* [e01a6c026d] Naveen Albert -- func_scramble: Audio scrambler function
ASTERISK-29478: Function to drop frames in the TX or RX directions
Reported by: N A
* [7383f74dfc] Naveen Albert -- func_frame_drop: New function
ASTERISK-29477: Function to asynchronously store digits dialed
Reported by: N A
* [6645cf8d45] Naveen Albert -- app_dtmfstore: New application to store
digits
ASTERISK-29431: Minimum and maximum dialplan functions
Reported by: N A
* [eeffad1b62] Naveen Albert -- func_math: Three new dialplan functions
Category: Functions/func_channel
ASTERISK-29656: Add CHANNEL_EXISTS function
Reported by: N A
* [f38c7d67d3] Naveen Albert -- func_channel: Add CHANNEL_EXISTS
function.
Category: Functions/func_env
ASTERISK-29628: Add file and directory functions
Reported by: N A
* [71b021433f] Naveen Albert -- func_env: Add DIRNAME and BASENAME
functions
Category: Functions/func_strings
ASTERISK-29627: Add STRBETWEEN function
Reported by: N A
* [d5a53efb4f] Naveen Albert -- func_strings: Add STRBETWEEN function
Category: Functions/func_volume
ASTERISK-29439: func_volume: Volume function can't be read
Reported by: N A
* [19b5097d87] Naveen Albert -- func_volume: Add read capability to
function.
Category: Resources/NewFeature
ASTERISK-29546: Add tone detection module
Reported by: N A
* [a94b51ee60] Naveen Albert -- res_tonedetect: Tone detection module
Category: Resources/res_pjsip_diversion
ASTERISK-29027: Implement support for History-Info
Reported by: Torrey Searle
* [888090ab18] Torrey Searle -- res_pjsip_diversion: implement support
for History-Info
Category: Resources/res_pjsip_header_funcs
ASTERISK-29389: Add PJSIP_HEADERS() and ability to read header by pattern
Reported by: Igor Goncharovsky
* [ac958b0f50] Igor Goncharovsky -- res_pjsip_header_funcs: Add
PJSIP_HEADERS() ability to read header by pattern
Bug
Category: . I did not set the category correctly.
ASTERISK-29146: GCC Warnings: ‘%s’ directive argument is null.
Reported by: Alexander Traud
* [28faafd1c4] Alexander Traud -- Compiler fixes for GCC when printf %s
is NULL
Category: Applications/General
ASTERISK-29287: app.h: C++ compatibility broken
Reported by: Jean Aunis - Prescom
* [725eca3bfa] Jaco Kroon -- app.h: Restore C++ compatibility for macro
AST_DECLARE_APP_ARGS
Category: Applications/app_agent_pool
ASTERISK-29614: app_agent_pool: XML Doc: unterminated entity reference
Reported by: Alexander Traud
* [16b0f460f6] Sean Bright -- config_options: Handle ACO arrays
correctly in generated XML docs.
Category: Applications/app_chanspy
ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
Reported by: Hendrik Wedhorn
* [357510cec3] Sean Bright -- app_chanspy: Spyee information missing in
ChanSpyStop AMI Event
Category: Applications/app_confbridge
ASTERISK-29618: ConfBridge errors on creation conference room
Reported by: Alexander Zharov
* [0070b9184c] George Joseph -- bridge_softmix: Suppress error on
topology change failure
ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
muting over AMI occurs
Reported by: Stefan Ruf
* [cc127a999c] Joshua C. Colp -- channel: Fix crash in suppress API.
* [3e5b9e3952] Joshua C. Colp -- channel: Fix memory leak in suppress
API.
Category: Applications/app_dial
ASTERISK-29329: app_dial: DTMF to 'D' option gets duplicated if there are
multiple progress events
Reported by: N A
* [8987de270f] Sean Bright -- app_dial.c: Only send DTMF on first
progress event.
Category: Applications/app_directory
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [57ee79a563] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Applications/app_milliwatt
ASTERISK-29575: app_milliwatt: Milliwatt application doesn't use the
proper timings
Reported by: N A
* [3f9ef427b5] Naveen Albert -- app_milliwatt: Timing fix
Category: Applications/app_mixmonitor
ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
Reported by: Robert Sutton
* [3bcf483373] Kevin Harwell -- app_mixmonitor: cleanup datastore when
monitor thread fails to launch
Category: Applications/app_mp3
ASTERISK-29635: MP3Player don' t work with actual mpg123 versions
Reported by: Carlos Oliva
* [ad1f7fae70] Carlos Oliva -- app_mp3: Force output to 16 bits in
mpg123
Category: Applications/app_page
ASTERISK-16799: Callee declined when 'beep' audio file does not exist
Reported by: IAMJames_
* [932eae69ab] Sean Bright -- app_page.c: Don't fail to Page if beep
sound file is missing
Category: Applications/app_playback
ASTERISK-27871: Remote URL in playback must end with file extension
Reported by: Caesar
* [d5bb27a06f] Sean Bright -- res_http_media_cache.c: Fix merge errors
from 18 -> master
* [d568326807] Sean Bright -- res_http_media_cache.c: Parse media URLs
to find extensions.
Category: Applications/app_queue
ASTERISK-29578: app_queue: Custom device state using included hints do not
update
Reported by: N A
* [eff78c8549] Naveen Albert -- app_queue: Fix hint updates for included
contexts
ASTERISK-28701: app_queue: Core reload resets queue stats, even when
keepstats=yes
Reported by: Luke Escude
* [9e947b0463] Naveen Albert -- app_queue: Don't reset queue stats on
reload
ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not
working
Reported by: Michael
* [8db2a34065] Sean Bright -- app_queue: Add alembic migration to add
ringinuse to queue_members.
ASTERISK-26614: app_queue: updatecdr option in queues.conf does
effectively nothing
Reported by: Alexander Gonchiy
* [aac442eecd] Sean Bright -- app_queue.c: Remove dead 'updatecdr' code.
ASTERISK-24631: Incorrect description of option "context" in
queues.conf.sample
Reported by: Etienne Lessard
* [cad843fe07] Sean Bright -- queues.conf.sample: Correct 'context'
documentation.
ASTERISK-27542: app_queue: When "queue show" CLI command is executed a
crash occurs
Reported by: Miguel Sanz
* [8d3d7bdb82] Sean Bright -- app_queue.c: Don't crash when realtime
queue name is empty.
ASTERISK-29355: app_queue: Queue member status message sent even if status
doesn't change
Reported by: Roman Pertsev
* [a8a08bcd1e] Joshua C. Colp -- app_queue: Only send QueueMemberStatus
if status changes.
ASTERISK-28369: app_queue: Member device state "invalid" when second call
is ringing and hint is used
Reported by: Boolah
* [4d8fc97e4a] Ivan Poddubnyi -- app_queue: Fix conversion of complex
extension states into device states
ASTERISK-29155: app_queue: Deadlock between queues container and
individual queues
Reported by: George Joseph
* [73f458b1e0] George Joseph -- app_queue: Fix deadlock between update
and show queues
ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove Aursand
* [c3a3ab8628] Kfir Itzhak -- app_queue: Fix leave-empty not recording a
call as abandoned
ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned
Reported by: Kfir Itzhak
* [c3a3ab8628] Kfir Itzhak -- app_queue: Fix leave-empty not recording a
call as abandoned
ASTERISK-29034: Lastpause of realtime members is reseting
Reported by: Evandro César Arruda
* [b2bd38a4f0] Evandro César Arruda -- app_queue: Member lastpause time
reseting
Category: Applications/app_read
ASTERISK-29673: app_read: Fix null pointer crash regression
Reported by: N A
* [60bbfe4572] Naveen Albert -- app_read: Fix null pointer crash
Category: Applications/app_saynumber
ASTERISK-29475: SayNumber triggers WARNING if caller hangs up during
application execution
Reported by: N A
* [f812c57477] Naveen Albert -- pbx_builtins: Corrects SayNumber warning
Category: Applications/app_skel
ASTERISK-29614: app_agent_pool: XML Doc: unterminated entity reference
Reported by: Alexander Traud
* [16b0f460f6] Sean Bright -- config_options: Handle ACO arrays
correctly in generated XML docs.
Category: Applications/app_voicemail
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [57ee79a563] Alexander Traud -- Compiler fixes for GCC with -Og
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [4b5ed817bd] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it
is not sent by email/processed by the mailcmd command
Reported by: Leandro Dardini
* [c925ed0eb9] Sean Bright -- app_voicemail: Process urgent messages
with mailcmd
Category: Bridges/bridge_simple
ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at
channel_internal_api.c:1590
Reported by: Ross Beer
* [44aef0449a] George Joseph -- bridge_channel_write_frame: Check for
NULL channel
ASTERISK-29161: Incorrect setup of recall channels
Reported by: Boris P. Korzun
* [8cb439f7e4] Boris P. Korzun -- bridge_basic: Fixed setup of recall
channels
Category: CDR/General
ASTERISK-29168: Asterisk crashes during call transfer
Reported by: Dalius Mockevicius
* [4274a4a7dd] Kevin Harwell -- pbx_realtime: wrong type stored on
publish of ast_channel_snapshot_type
Category: CDR/cdr_adaptive_odbc
ASTERISK-29494: cdr_adaptive_odbc: Prevent throwing warnings if CDR
filtering is used
Reported by: N A
* [4c49c84dee] Naveen Albert -- cdr_adaptive_odbc: Prevent filter
warnings
Category: Channels/General
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [57ee79a563] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Channels/chan_dahdi
ASTERISK-29518: sig_analog: FCG_CAMA fails to signal ANI spill when using
MF signaling
Reported by: Sarah Autumn
* [db4a3b117d] Sarah Autumn -- sig_analog: Changes to improve
electromechanical signalling compatibility
Category: Channels/chan_iax2
ASTERISK-20219: [patch] - IAX2 Call Encryption Fails with RSA
authentication
Reported by: Michael Munger
* [32ea7c7ca5] Naveen Albert -- chan_iax2: Add encryption for RSA
authentication
ASTERISK-29392: chan_iax2: Asterisk crashes when queueing video with
format
Reported by: Michael Welk
* [56f9c28a50] Kevin Harwell -- AST-2021-008 - chan_iax2: remote crash
on unsupported media format
Category: Channels/chan_local
ASTERISK-29407: chan_local: Filtering audio formats should not occur on
removed streams
Reported by: Joshua C. Colp
* [f142ca254e] Joshua C. Colp -- chan_local: Skip filtering audio
formats on removed streams.
ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing
Reported by: Matthias Hensler
* [970b84946e] Joshua C. Colp -- core_unreal: Fix deadlock with T.38
control frames.
* [00b229c69c] Ben Ford -- core_unreal: Fix T.38 faxing when using local
channels.
Category: Channels/chan_mgcp
ASTERISK-20339: chan_mgcp, resp_pktccops ast_debug support
Reported by: Tomas Maldonado
* [41ed46f474] Sean Bright -- mgcp: Remove dead debug code
Category: Channels/chan_pjsip
ASTERISK-28393: Multidomain support issue
Reported by: Andrea Sannucci
* [98e4119642] Joseph Nadiv -- res_pjsip.c: Support endpoints with
domain info in username
ASTERISK-29358: chan_pjsip: Trace message for progress is output even if
frame is not queued
Reported by: Michael Maier
* [1b41629447] Sean Bright -- chan_pjsip: Correct misleading trace
message
ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
instead of a channel variable
Reported by: Ivan Poddubny
* [f2aa6c7017] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
creating a channel
ASTERISK-27902: chan_pjsip isn't updating hangupcause on 4XX responses
Reported by: George Joseph
* [134d2e729d] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
Reported by: Alex Hermann
* [134d2e729d] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
Reported by: Julien
* [134d2e729d] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if
registration can't be send
Reported by: Michael Maier
* [9a4486e9fb] George Joseph -- Revert
"res_pjsip_outbound_registration.c: Use our own scheduler and other
stuff"
ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the
Transfer result
Reported by: Dan Cropp
* [ffa87ecade] Dan Cropp -- chan_pjsip: Incorporate channel reference
count into transfer_refer().
ASTERISK-29210: res_pjsip: Crash when examining transport
Reported by: N GM
* [505939c9ed] Nick French -- res_pjsip: Prevent segfault in UDP
registration with flow transports
ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts
Reported by: Sean Bright
* [6475fe3dd7] Joshua C. Colp -- pjsip: Match lifetime of INVITE session
to our session.
ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
Reported by: Joseph Ades
* [3c4a1722b6] Kevin Harwell -- chan_pjsip: disallow
PJSIP_SEND_SESSION_REFRESH pre-answer execution
Category: Channels/chan_sip/CodecHandling
ASTERISK-29280: chan_sip: Allow peers without audio (text+video).
Reported by: Alexander Traud
* [1f77c33c02] Alexander Traud -- chan_sip: Allow [peer] without audio
(text+video).
ASTERISK-29265: chan_sip: Allow text+video media streams, again.
Reported by: Alexander Traud
* [620d9f4782] Alexander Traud -- chan_sip: Set up calls without audio
(text+video), again.
ASTERISK-29258: chan_sip: Audio stream rejected, Other stream present:
Invalid SDP.
Reported by: Alexander Traud
* [4aff42b274] Alexander Traud -- chan_sip: SDP: Reject audio streams
correctly.
ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
accepted.
Reported by: Alexander Traud
* [1c05667cfc] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
* [1c05667cfc] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Channels/chan_sip/General
ASTERISK-29370: chan_sip does not recognize application/hook-flash
Reported by: N A
* [fd40752954] Naveen Albert -- chan_sip: Expand hook flash recognition.
ASTERISK-29030: res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
gets inserted when switching from progress to established
Reported by: Matthias Hensler
* [b1807d440e] Sean Bright -- res_rtp_asterisk: More robust timestamp
checking
ASTERISK-29011: chan_sip: ToHost property not cleared on reload
Reported by: Dennis
* [aab666bb9d] Dennis Buteyn -- chan_sip: Clear ToHost property on peer
when changing to dynamic host
Category: Channels/chan_sip/SRTP
ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled
user-agent.
Reported by: Alexander Traud
* [1c05667cfc] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Channels/chan_sip/TCP-TLS
ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
Reported by: Alexander Traud
* [103d7da3bb] Alexander Traud -- chan_sip: Remove unused
sip_socket->port.
Category: Channels/chan_sip/Video
ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are
accepted.
Reported by: Alexander Traud
* [1c05667cfc] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
Reported by: Alexander Traud
* [1c05667cfc] Alexander Traud -- chan_sip: SDP: Sidestep stream parsing
when its media is disabled.
Category: Configs/Samples
ASTERISK-29123: logger.conf.sample missing comment mark on line 115
Reported by: Andrew Siplas
* [0190e706b8] Andrew Siplas -- logger.conf.sample: add missing comment
mark
Category: Contrib/General
ASTERISK-29142: sip_to_pjsip.py: doesn't read globbed includes
Reported by: Michael Newton
* [a5d55fc9e1] Sean Bright -- sip_to_pjsip.py: Handle #include globs and
other fixes
Category: Core/ACL
ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault
on reading rule from realtime
Reported by: Andrew Yager
* [c3588d9c0b] Sean Bright -- acl.c: Coerce a NULL pointer into the
empty string
Category: Core/Bridging
ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and
muting over AMI occurs
Reported by: Stefan Ruf
* [cc127a999c] Joshua C. Colp -- channel: Fix crash in suppress API.
* [3e5b9e3952] Joshua C. Colp -- channel: Fix memory leak in suppress
API.
Category: Core/BuildSystem
ASTERISK-29348: menuselect doesn't return errors in many cases
Reported by: George Joseph
* [fc03116d9b] Jaco Kroon -- menuselect: exit non-zero in case of
failure on --enable|disable options.
Category: Core/Channels
ASTERISK-29259: channel: Allow text+video media streams, again.
Reported by: Alexander Traud
* [6d980de282] Alexander Traud -- channel: Set up calls without audio
(text+video), again.
ASTERISK-29091: Crash when ast_translator_build_path fails
Reported by: Jasper van der Neut
* [e831952eba] Jasper van der Neut -- channels: Don't dereference NULL
pointer
Category: Core/CodecInterface
ASTERISK-29526: G729 audio gets corrupted by Asterisk due to smoother
Reported by: under
* [ff8ca2c9f1] under -- codec_builtin.c: G729 audio gets corrupted by
Asterisk due to smoother
ASTERISK-29328: translate.c: possible buffer overflow when upsampling
Reported by: Jean Aunis - Prescom
* [55279bfd9c] Jean Aunis -- translate.c: Take sampling rate into
account when checking codec's buffer size
Category: Core/DNS
ASTERISK-28004: dns: Core ast_dns_get_nameservers does not support
configured IPv6 servers
Reported by: Isaac McDonald
* [5a5ea06ffc] Sean Bright -- dns.c: Load IPv6 DNS resolvers if
configured.
Category: Core/General
ASTERISK-12: app_voicemail2 became a bit silent, lately
Reported by: siggi
* [ff8ca2c9f1] under -- codec_builtin.c: G729 audio gets corrupted by
Asterisk due to smoother
ASTERISK-29372: file.c switch does not account for flash events
Reported by: N A
* [0026aeada3] Naveen Albert -- main/file.c: Don't throw error on flash
event.
ASTERISK-29306: strings: Incorrect use of __attribute__((pure)) in
ast_str_to_lower definition
Reported by: Vitezslav Novy
* [30e509c2f9] Sean Bright -- strings.h: ast_str_to_upper() and
_to_lower() are not pure.
ASTERISK-28430: res_rtp_asterisk.c: FRACK!, Failed assertion errno !=
EBADF
Reported by: under
* [fa023cbfa0] Sean Bright -- tcptls.c: Don't close TCP client file
descriptors more than once
ASTERISK-28311: dsp: ast_dsp_silence_noise_with_energy wrong judgment of
frame format
Reported by: 周家建
* [16dfe8f03f] Sean Bright -- dsp.c: Update calls to ast_format_cmp to
check result properly
Category: Core/Internationalization
ASTERISK-29297: say: Y2021 problem – Asterisk cannot say year 2021 in
Dutch
Reported by: Jacek Konieczny
* [2ea75ed3d5] Nico Kooijman -- main: With Dutch language year after
2020 is not spoken in say.c
Category: Core/Jitterbuffer
ASTERISK-27176: test_abstract_jb: frames leak
Reported by: Corey Farrell
* [085cc94f16] Sean Bright -- test_abstract_jb.c: Fix put and
put_out_of_order memory leaks.
ASTERISK-29480: fixedjitterbuffer contains an un-wrappered assert that
triggers on a negative time slew
Reported by: Dan Cropp
* [bc973bd719] George Joseph -- jitterbuffer: Correct signed/unsigned
mismatch causing assert
Category: Core/Logging
ASTERISK-29209: Debug messages printed by scope trace might be missing
newlines
Reported by: Alexander Traud
* [7d4ae7dc18] George Joseph -- logger.c: Automatically add a newline to
formats that don't have one
Category: Core/PBX
ASTERISK-29485: core: Inband generation of tones for Busy() and
Congestion() may not occur
Reported by: Joshua C. Colp
* [5382b9dbb8] Joshua C. Colp -- core: Don't play silence for Busy() and
Congestion() applications.
ASTERISK-29441: Core reload making TCP endpoints go offline
Reported by: Luke Escude
* [44fde9f428] Joshua C. Colp -- res_pjsip: On partial transport reload
also move factories.
Category: Core/RTP
ASTERISK-28416: Unable to get rtp codec payload code for slin
Reported by: Brian J. Murrell
* [30e08ce1bb] Sean Bright -- format_cap: Perform codec lookups by
pointer instead of name
Category: Core/Stasis
ASTERISK-28237: "FRACK!, Failed assertion bad magic number" happens when
unsubscribe an application from an event source
Reported by: Lucas Tardioli Silveira
* [2193cf1b26] Evgenios_Greek -- stasis: Fix "FRACK!, Failed assertion
bad magic number" when unsubscribing
ASTERISK-29355: app_queue: Queue member status message sent even if status
doesn't change
Reported by: Roman Pertsev
* [a8a08bcd1e] Joshua C. Colp -- app_queue: Only send QueueMemberStatus
if status changes.
Category: Documentation
ASTERISK-29614: app_agent_pool: XML Doc: unterminated entity reference
Reported by: Alexander Traud
* [16b0f460f6] Sean Bright -- config_options: Handle ACO arrays
correctly in generated XML docs.
ASTERISK-24434: Fix differing usage of assignment operators in
modules.conf
Reported by: Rusty Newton
* [c2dbfb9a8e] Sean Bright -- modules.conf: Fix more differing usages of
assignment operators.
* [55bd104589] Sean Bright -- modules.conf: Fix differing usage of
assignment operators.
ASTERISK-24631: Incorrect description of option "context" in
queues.conf.sample
Reported by: Etienne Lessard
* [cad843fe07] Sean Bright -- queues.conf.sample: Correct 'context'
documentation.
ASTERISK-25358: dateformat not read from logger.conf by remote console
Reported by: Igor Liferenko
* [b4347c4861] Mark Murawski -- logger: Console sessions will now
respect logger.conf dateformat= option
ASTERISK-29136: config: Sample features.conf incorrectly includes " around
sound files
Reported by: Benjamin M.
* [8f33e23dfb] Sean Bright -- features.conf.sample: Sample sound files
incorrectly quoted
ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
Reported by: Eric Smith
* [4b5ed817bd] Sean Bright -- app_voicemail.c: Document VMSayName
interruption behavior
Category: Formats/General
ASTERISK-29539: Segmentation fault at ast_writestream() when write handler
not defined (happens with OGG/Speex)
Reported by: Ernani José Camargo Azevedo
* [37f7d19c8c] Kevin Harwell -- format_ogg_speex: Implement a "not
supported" write handler
Category: Functions/func_curl
ASTERISK-28825: Any curl response checks out as valid even if 404 is
returned.
Reported by: dovid
* [bc58e84f47] Dovid Bender -- func_curl.c: Allow user to set what
return codes constitute a failure.
ASTERISK-29085: func_curl: Segmentation fault when using CURL after
setting httpheader CURLOPT
Reported by: Péter Juhász
* [b11b49945b] Sean Bright -- func_curl.c: Prevent crash when using
CURLOPT(httpheader)
Category: Functions/func_lock
ASTERISK-29217: LOCK() can grant the same lock to multiple channels
spuriously
Reported by: Jaco Kroon
* [c797500956] Jaco Kroon -- func_lock: fix multiple-channel-grant
problems.
Category: Functions/func_odbc
ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
Reported by: Alexander Traud
* [57ee79a563] Alexander Traud -- Compiler fixes for GCC with -Og
Category: Functions/func_version
ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
versions
Reported by: cmaj
* [3040edcbb1] cmaj -- Makefile: Fix certified version numbers
Category: General
ASTERISK-29630: Asterisk is unable to read extended number format terminfo
files
Reported by: Sean Bright
* [61136fd297] Sean Bright -- term.c: Add support for extended number
format terminfo files.
ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend
Reported by: Alexander Traud
* [b91fb3c396] Alexander Traud -- loader: Sync load- and build-time
deps.
Category: PBX/General
ASTERISK-29046: pbx: Deadlock when doing a reload, while simultaneously
doing an ExtensionState on a pattern match hint that ends up adding an
extension
Reported by: Ramarajan
* [28bae5e901] Joshua C. Colp -- pbx: Fix hints deadlock between reload
and ExtensionState.
Category: PBX/pbx_ael
ASTERISK-29609: Subsequent 'ael reload' will cause a lock up
Reported by: Mark Murawski
* [185321066f] Mark Murawski -- pbx_ael: Fix crash and lockup issue
regarding 'ael reload'
Category: Resources/General
ASTERISK-29130: prometheus: Crash when scraping bridge
Reported by: Francisco Correia
* [53c702e1cc] George Joseph -- res_prometheus: Clone containers before
iterating
Category: Resources/res_ari_bridges
ASTERISK-29668: ari: Listing bridges fails when dialing bridge exists
Reported by: Joshua C. Colp
* [35a94ec708] Joshua C. Colp -- ari: Ignore invisible bridges when
listing bridges.
Category: Resources/res_ari_channels
ASTERISK-29629: ARI external media channel creation doesn't set option
data
Reported by: sungtae kim
* [4d9ba65c53] Sungtae Kim -- resource_channels.c: Fix external media
data option
ASTERISK-29622: ARI: external media create doesn't use body parameter
Reported by: sungtae kim
* [3c31b6aaa2] sungtae kim -- resource_channels.c: Fix wrong external
media parameter parse
ASTERISK-29514: ari: Audiosocket segfault when no data specified
Reported by: Igor Goncharovsky
* [99d44f0c5a] Igor Goncharovsky -- res_ari: Fix audiosocket segfault
ASTERISK-29188: null media causing the Asterisk crash
Reported by: sungtae kim
* [91fc57f56b] Sungtae Kim -- res_ari: Fix wrong media uri handle for
channel play
Category: Resources/res_ari_endpoints
ASTERISK-29108: resource_endpoints.c : Memory leak if endpoint not found
Reported by: Jean Aunis - Prescom
* [61116d5dbc] Jean Aunis -- resource_endpoints.c: memory leak when
providing a 404 response
Category: Resources/res_config_pgsql
ASTERISK-29293: res_config_pgsql: Limit realtime_pgsql() to return one (no
more) record
Reported by: Boris P. Korzun
* [b046e960af] Boris P. Korzun -- res_config_pgsql: Limit
realtime_pgsql() to return one (no more) record.
Category: Resources/res_convert
ASTERISK-29539: Segmentation fault at ast_writestream() when write handler
not defined (happens with OGG/Speex)
Reported by: Ernani José Camargo Azevedo
* [37f7d19c8c] Kevin Harwell -- format_ogg_speex: Implement a "not
supported" write handler
Category: Resources/res_fax
ASTERISK-29312: res_fax: asterisk fails to publish the Stasis and
ReceiveFax status messages if the remote Station ID contains invalid UTF-8
characters
Reported by: Alexei Gradinari
* [d2f623bae2] Alexei Gradinari -- res_fax: validate the remote/local
Station ID for UTF-8 format
Category: Resources/res_http_media_cache
ASTERISK-27871: Remote URL in playback must end with file extension
Reported by: Caesar
* [d5bb27a06f] Sean Bright -- res_http_media_cache.c: Fix merge errors
from 18 -> master
* [d568326807] Sean Bright -- res_http_media_cache.c: Parse media URLs
to find extensions.
ASTERISK-29173: Media cache URL requests allow infinite redirects
Reported by: Sean Bright
* [90fd1fd96a] Sean Bright -- res_http_media_cache.c: Set reasonable
number of redirects
Category: Resources/res_musiconhold
ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold
without entries
Reported by: Nathan Bruning
* [5e426987c2] Nathan Bruning -- res_musiconhold: Don't crash when
real-time doesn't return any entries
ASTERISK-29099: res_musiconhold: Realtime MOH only loads a single entry
Reported by: laszlovl
* [990c72bbcf] laszlovl -- res_musiconhold: Load all realtime entries,
not just the first
ASTERISK-24329: Music On Hold announcement cuts intro of music the first
time it is played
Reported by: Thomas Frederiksen
* [0aaf9aa6de] Sean Bright -- res_musiconhold: Start playlist after
initial announcement
ASTERISK-28927: Asterisk crash in music on hold
Reported by: David Cunningham
* [b7c2205402] Sean Bright -- res_musiconhold.c: Prevent crash with
realtime MoH
Category: Resources/res_odbc
ASTERISK-29311: res_odbc_transaction sets forcecommit default value based
on isolation level instead of forcecommit
Reported by: Jaco Kroon
* [6d2614be68] Jaco Kroon -- res_odbc_transaction: correctly initialise
forcecommit value from DSN.
Category: Resources/res_parking
ASTERISK-29042: res_parking: Parker UUID is no longer copied
Reported by: Misha Vodsedalek
* [c4bed96742] Joshua C. Colp -- parking: Copy parker UUID as well.
Category: Resources/res_pjproject
ASTERISK-29582: res_pjproject: Can't map pjproject log messages to
Asterisk TRACE
Reported by: George Joseph
* [a662d75556] George Joseph -- res_pjproject: Allow mapping to Asterisk
TRACE level
Category: Resources/res_pjsip
ASTERISK-29618: ConfBridge errors on creation conference room
Reported by: Alexander Zharov
* [0070b9184c] George Joseph -- bridge_softmix: Suppress error on
topology change failure
ASTERISK-29354: res_pjsip: Allow partial reloading of transports
Reported by: Joshua C. Colp
* [71dfbdc7b9] Joshua C. Colp -- res_pjsip: Add support for partial
transport reload.
ASTERISK-29196: res_pjsip: Segmentation fault
Reported by: Mauri de Souza Meneguzzo (3CPlus)
* [492945ac60] Joshua C. Colp -- pjsip: Make modify_local_offer2
tolerate previous failed SDP.
ASTERISK-29261: res_pjsip: user=phone validation fail for isup numbers
containing *#
Reported by: Mark Petersen
* [9b5d20e3d5] Mark Petersen -- res/res_pjsip.c: allow user=phone when
number contain *#
ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS
response
Reported by: Alexander Greiner-Baer
* [fba10fb54c] Alexander Greiner-Baer -- res_pjsip: set Accept-Encoding
to identity in OPTIONS response
ASTERISK-28933: res_pjsip.so fails to load when bundled pjproject is
compiled without libssl
Reported by: Walter Doekes
* [b52acb87b0] Alexander Traud -- res_pjsip/config_transport: Load and
run without OpenSSL.
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with
auth) on 407 replies
Reported by: Sebastian Damm
* [cd8f8b94f8] Ben Ford -- AST-2020-002 - res_pjsip: Stop sending
INVITEs after challenge limit.
ASTERISK-29124: res_pjsip: flow transport broken for outbound requests
Reported by: Nick French
* [bd98e153d1] Nick French -- res_pjsip_session: Restore calls to
ast_sip_message_apply_transport()
ASTERISK-28995: res_pjsip_registrar: Expires on statically configured
contacts is not correct
Reported by: tootai
* [921b1a02c4] Joshua C. Colp -- res_pjsip_registrar: Don't specify an
expiration for static contacts.
Category: Resources/res_pjsip/Bundling
ASTERISK-29654: pjproject includes trailing whitespace in sdp format
attributes
Reported by: George Joseph
* [3d6e133ccf] George Joseph -- pjproject: Add patch to fix trailing
whitespace issue in rtpmap
Category: Resources/res_pjsip_authenticator_digest
ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with
auth) on 407 replies
Reported by: Sebastian Damm
* [cd8f8b94f8] Ben Ford -- AST-2020-002 - res_pjsip: Stop sending
INVITEs after challenge limit.
Category: Resources/res_pjsip_config_wizard
ASTERISK-29503: Updated identify/match syntax not supported by config
wizard
Reported by: Sean Bright
* [0ac9c83561] Sean Bright -- res_pjsip_config_wizard.c: Add port
matching support.
ASTERISK-29097: res_pjsip_config_wizard: Crash when freeing string when
failing to add extension
Reported by: Vieri
* [51cba591e3] Sean Bright -- pbx.c: On error,
ast_add_extension2_lockopt should always free 'data'
Category: Resources/res_pjsip_diversion
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [51e2187a14] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
ASTERISK-29001: chan_pjsip does not process or forward 181 responses
Reported by: Torrey Searle
* [04051b324b] Torrey Searle -- res_pjsip_diversion: handle 181
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-29503: Updated identify/match syntax not supported by config
wizard
Reported by: Sean Bright
* [0ac9c83561] Sean Bright -- res_pjsip_config_wizard.c: Add port
matching support.
Category: Resources/res_pjsip_messaging
ASTERISK-29663: messaging: AMI MessageSend does not support same
parameters as dialplan application
Reported by: Brian J. Murrell
* [52b5821694] Sean Bright -- message.c: Support 'To' header override
with AMI's MessageSend.
ASTERISK-29404: Consolidate res_pjsip_messaging fixes for domain name
Reported by: George Joseph
* [c3654a9959] George Joseph -- res_pjsip_messaging: Refactor outgoing
URI processing
Category: Resources/res_pjsip_nat
ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
with external_signaling_address
Reported by: Brian Paboojian
* [2c1b6b7b15] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
REGISTER responses.
Category: Resources/res_pjsip_outbound_authenticator_digest
ASTERISK-29397: pjsip: Asterisk isn't tolerant of RFC8760 UASs
Reported by: George Joseph
* [9cc1d6fc22] George Joseph -- res_pjsip_outbound_authenticator_digest:
Be tolerant of RFC8760 UASs
Category: Resources/res_pjsip_outbound_registration
ASTERISK-29315: res_pjsip: re-registration gets stuck if setting initial
auth credentials fails
Reported by: Nick French
* [8f6e0f9367] Nick French -- res_pjsip: dont return early from
registration if init auth fails
ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered
Reported by: Michael Maier
* [9a4486e9fb] George Joseph -- Revert
"res_pjsip_outbound_registration.c: Use our own scheduler and other
stuff"
Category: Resources/res_pjsip_refer
ASTERISK-29313: res_pjsip_refer: Segfault in progress notify
Reported by: George Joseph
* [4c9c5c985b] George Joseph -- res_pjsip_refer: Refactor progress
locking and serialization
Category: Resources/res_pjsip_registrar
ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses
with external_signaling_address
Reported by: Brian Paboojian
* [2c1b6b7b15] Joshua C. Colp -- res_pjsip_nat: Don't rewrite Contact on
REGISTER responses.
ASTERISK-28995: res_pjsip_registrar: Expires on statically configured
contacts is not correct
Reported by: tootai
* [921b1a02c4] Joshua C. Colp -- res_pjsip_registrar: Don't specify an
expiration for static contacts.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-29479: [patch] Channels are not put on hold for Session Progress
with inactive audio
Reported by: Bernd Zobl
* [c30f68a57b] Bernd Zobl -- res_pjsip_sdp_rtp: Evaluate remotely held
for Session Progress
ASTERISK-29105: chan_pjsip: 180 Ringing with SDP not changed into progress
Reported by: Sebastian Damm
* [48ed4f670f] Holger Hans Peter Freyther -- pjsip: Generate progress
(once) when receiving a 180 with a SDP
ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
on reinvite without SDP offer
Reported by: Michael Maier
* [a81d07ea56] Joshua C. Colp -- res_pjsip_session: Always produce offer
on re-INVITE without SDP.
ASTERISK-29051: res_pjsip_sdp_rtp: Does not set correct values on RTP
instance when "auto" DTMF is used
Reported by: Sebastian Damm
* [9c0ded6e76] Holger Hans Peter Freyther -- res_pjsip_sdp_rtp: Fix
accidentally native bridging calls
Category: Resources/res_pjsip_session
ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused
asterisk crash
Reported by: sungtae kim
* [a03a05195a] George Joseph -- res_pjsip_session: Make
reschedule_reinvite check for NULL topologies
* [02c4b2ac60] Sungtae Kim -- res_pjsip_session: Fixed NULL active media
topology handle
ASTERISK-29303: pjsip: Re-invite occurs when it shouldn't
Reported by: Benjamin Keith Ford
* [e1126ffc10] Ben Ford -- res_pjsip_session.c: Check topology on
re-invite.
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [5e998d8bd3] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
ASTERISK-29248: res_pjsip_session: res sometimes uninitialized reported by
compiler Clang.
Reported by: Alexander Traud
* [df6afadf26] Alexander Traud -- res_pjsip_session: Avoid
sometimes-uninitialized warning with Clang.
ASTERISK-29220: After T38 reinvite response of 488 a subsequent G711
reinvite is not processed correctly. Instead the previous T38 session
media is used
Reported by: Robert Cripps
* [24e678b9bb] Robert Cripps -- res/res_pjsip_session.c: Check that
media type matches in
ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
instead of a channel variable
Reported by: Ivan Poddubny
* [f2aa6c7017] Ivan Poddubnyi -- chan_pjsip: Assign SIPDOMAIN after
creating a channel
ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due
to codec negotiation after upgrading from Asterisk 16
Reported by: Ross Beer
* [dcd2ed69a3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call
pref.
ASTERISK-29014: res_pjsip_session: Re-INVITE collisions aren't handled
correctly
Reported by: George Joseph
* [53910b1f25] George Joseph -- res_pjsip_session: Fix issue with COLP
and 491
* [86f1bce186] George Joseph -- res_pjsip_session: Handle multi-stream
re-invites better
ASTERISK-29033: res_pjsip_session: Aggressively terminates session on
failed re-INVITE
Reported by: Joshua C. Colp
* [71ceefa75d] Joshua C. Colp -- res_pjsip_session: Don't aggressively
terminate on failed re-INVITE.
Category: Resources/res_pjsip_t38
ASTERISK-29402: res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
does not support it
Reported by: Matthew Kern
* [9d04535bbd] Matthew Kern -- res_pjsip_t38: bind UDPTL sessions like
RTP
ASTERISK-29203: res_pjsip_t38: Crash when changing state
Reported by: Gregory Massel
* [5e998d8bd3] Kevin Harwell -- AST-2021-002: Remote crash possible when
negotiating T.38
Category: Resources/res_rtp_asterisk
ASTERISK-29671: res_rtp_asterisk: memory leak
Reported by: Jean Aunis - Prescom
* [576119e076] Jean Aunis -- res_rtp_asterisk: fix memory leak
ASTERISK-29660: Build failure when disabling PJSIP support
Reported by: Guido Falsi
* [675adbf0f5] Guido Falsi -- res_rtp_asterisk.c: Fix build failure when
not building with pjproject.
ASTERISK-29616: res_rtp_asterisk: sqrt(.) requires the header math.h.
Reported by: Alexander Traud
* [e65e1c5c6c] Alexander Traud -- res_rtp_asterisk: sqrt(.) requires the
header math.h.
ASTERISK-29507: STUN timeout is silently delaying calls
Reported by: Sébastien Duthil
* [8a21d466ea] Sebastien Duthil -- stun: Emit warning message when STUN
request times out
ASTERISK-29433: res_rtp_asterisk: Server reflexive candidates use
incorrect raddr for RTCP
Reported by: Chris
* [a985e5069c] Joshua C. Colp -- res_rtp_asterisk: Set correct raddr
port on RTCP srflx candidates.
ASTERISK-29030: res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
gets inserted when switching from progress to established
Reported by: Matthias Hensler
* [b1807d440e] Sean Bright -- res_rtp_asterisk: More robust timestamp
checking
ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation
Reported by: Kevin Harwell
* [0fc906a5e1] Kevin Harwell -- res_rtp_asterisk: Fix standard deviation
calculation
ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated
Reported by: N A
* [8bd13a995a] Joshua C. Colp -- res_rtp_asterisk: Only raise flash
control frame on end.
ASTERISK-29352: res_rtp_asterisk: Fix frame delivery time when SSRC
changes
Reported by: Joshua C. Colp
* [cce5ee5b7a] Joshua C. Colp -- res_rtp_asterisk: Force resync on SSRC
change.
ASTERISK-29300: res_rtp_asterisk: When native local bridging the remote
SSRC becomes permanent
Reported by: Sebastian Damm
* [8c247e2a94] Torrey Searle -- res/res_rtp_asterisk: generate new SSRC
on native bridge end
ASTERISK-29266: ICE Role conflict with an unauthorized session
Reported by: Salah Ahmed
* [5d42dd2e6a] Salah Ahmed -- res_rtp_asterisk: Check remote ICE reset
and reset local ice attrb
ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold
from webrtc client
Reported by: Edvin Vidmar
* [e7b13df394] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch
that leads to overflow
ASTERISK-29089: RTP Ports not cleared after hangup
Reported by: Ross Beer
* [f67f5676b7] Joshua C. Colp -- res_pjsip_session: Fix session
reference leak.
ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string
to each message block.
Reported by: Thomas Johnson
* [3553192900] Sean Bright -- bridge_channel: Ensure text messages are
zero terminated
Category: Resources/res_snmp
ASTERISK-29634: res_snmp: gcc 11 needs -fPIC to compile correctly
Reported by: George Joseph
* [df63a99337] George Joseph -- res_snmp: Add -fPIC to _ASTCFLAGS
Category: Resources/res_speech
ASTERISK-29040: res_speech: Assertion on format
Reported by: Nickolay V. Shmyrev
* [5b9ac90531] Nickolay Shmyrev -- res_speech: Bump reference on format
object
Category: Resources/res_stasis
ASTERISK-29229: Stasis/messaging: text messages not dispatched to all
subscribers when using generic subscription
Reported by: Jean Aunis - Prescom
* [c559667868] Jean Aunis -- Stasis/messaging: tech subscriptions
conflict with endpoint subscriptions.
ASTERISK-29081: res_stasis: Add compare function for bridges moh container
Reported by: Hajek Michal
* [b4ab0dd41a] Michal Hajek -- res_stasis.c: Add compare function for
bridges moh container
ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info
Reported by: sungtae kim
* [c10ed8d4d6] sungtae kim -- stasis_bridge.c: Fixed wrong video_mode
shown
Category: Resources/res_statsd
ASTERISK-29513: statsd: Remove non-standard metric type Meter
Reported by: Rijnhard Hessel
* [f13eef719c] Rijnhard Hessel -- res_statsd: handle non-standard meter
type safely
Category: Resources/res_stir_shaken
ASTERISK-29175: res_pjsip_stir_shaken: Fix module description
Reported by: Stanislav Abramenkov
* [ab7a08b4ef] Stanislav -- res_pjsip_stir_shaken: Fix module
description
Category: Tests/General
ASTERISK-27176: test_abstract_jb: frames leak
Reported by: Corey Farrell
* [085cc94f16] Sean Bright -- test_abstract_jb.c: Fix put and
put_out_of_order memory leaks.
Category: Utilities/aelparse
ASTERISK-29540: aelparse: include of context with timings fails
Reported by: Alexander Traud
* [835ab50724] Alexander Traud -- aelparse: Accept an included context
with timings.
Category: Utilities/muted
ASTERISK-29145: GCC Warnings with OPTIMIZE=-Os make
Reported by: Alexander Traud
* [914aecb8d8] Alexander Traud -- Compiler fixes for GCC with -Os
Category: pjproject/pjsip
ASTERISK-24601: [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
event: dialog XML body
Reported by: Marco Paland
* [3cccdf6d98] Joseph Nadiv -- res_pjsip_dialog_info_body_generator: Add
LOCAL/REMOTE tags in dialog-info+xml
ASTERISK-29377: cpool_release_pool "double free or corruption (out)"
Reported by: Robert Sutton
* [49c2e7e307] Joshua C. Colp -- pjsip: Add patch for resolving STUN
packet lifetime issues.
ASTERISK-28452: pjsip: of SDP is not incremented though SDP may be changed
on reinvite without SDP offer
Reported by: Michael Maier
* [a81d07ea56] Joshua C. Colp -- res_pjsip_session: Always produce offer
on re-INVITE without SDP.
ASTERISK-29191: tel: URI in Diversion header causes crash
Reported by: Mikhail Ivanov
* [51e2187a14] Torrey Searle -- res/res_pjsip_diversion: prevent crash
on tel: uri in History-Info
ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set
Reported by: Flole Systems
* [0b10995811] Pirmin Walthert -- res_pjsip_nat.c: Create deep copies of
strings when appropriate
ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered
INVITE when NAT is active (UDP transport with external_media_address)
Reported by: Michael Neuhauser
* [e8c2ce2873] Michael Neuhauser -- pjproject: clone sdp to protect
against (nat) modifications
Improvement
Category: Applications/General
ASTERISK-29637: Add support for future dates in Say.c
Reported by: Shloime Rosenblum
* [f3ff893310] Shloime Rosenblum -- main/say.c: Support future dates
with Q and q format params
Category: Applications/app_mixmonitor
ASTERISK-29244: Add MixMonitorStart / Stop / Mute AMI events
Reported by: Sébastien Duthil
* [6e695c867f] Sebastien Duthil -- app_mixmonitor: Add AMI events
MixMonitorStart, -Stop and -Mute.
Category: Applications/app_morsecode
ASTERISK-29541: app_morsecode: Add American Morse code
Reported by: N A
* [b5044586f7] Naveen Albert -- app_morsecode: Add American Morse code
Category: Applications/app_originate
ASTERISK-29543: app_originate: Allow specifying codec(s) to use
Reported by: N A
* [2394757e55] Naveen Albert -- app_originate: Add ability to set codecs
ASTERISK-29450: Allow setting channel variables using Originate
application
Reported by: N A
* [b742514553] Naveen Albert -- app_originate: Allow setting Caller ID
and variables
Category: Applications/app_queue
ASTERISK-29528: Add support for multiple files for agent announcements
Reported by: N A
* [0975cff6c0] Naveen Albert -- app_queue: Allow streaming multiple
announcement files
Category: Applications/app_stack
ASTERISK-29626: app_stack: Include calling location if attempting to
branch to nonexistent location
Reported by: N A
* [5fe3a745e4] Naveen Albert -- app_stack: Include current location if
branch fails
Category: Applications/app_transfer
ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
failure SIP code
Reported by: Dan Cropp
* [55891227e8] Dan Cropp -- chan_pjsip, app_transfer: Add
TRANSFERSTATUSPROTOCOL variable
Category: Applications/app_voicemail
ASTERISK-29632: Add option to Application_VoiceMail to suppress
instructions only when a custom greeting is present
Reported by: Charlie Smurthwaite
* [f67b72093e] Sean Bright -- app_voicemail.c: Ability to silence
instructions if greeting is present.
ASTERISK-29349: Silent voicemail option is not completely silent
Reported by: N A
* [567ea5abf8] Naveen Albert -- app_voicemail: Configurable voicemail
beep
Category: Applications/app_voicemail/NewFeature
ASTERISK-29118: VoiceMail() should have an option to play greetings as
Early Media
Reported by: Juan Carlos Castro y Castro
* [eda3679c1c] Joshua C. Colp -- voicemail: add option 'e' to play
greetings as early media
Category: Channels/General
ASTERISK-29380: Add Flash AMI event to handle flash events
Reported by: N A
* [04454fc238] Naveen Albert -- AMI: Add AMI event to expose hook flash
events
Category: Channels/NewFeature
ASTERISK-29380: Add Flash AMI event to handle flash events
Reported by: N A
* [04454fc238] Naveen Albert -- AMI: Add AMI event to expose hook flash
events
Category: Channels/chan_iax2
ASTERISK-29605: chan_iax2: Add ANI2
Reported by: N A
* [29770520b3] Naveen Albert -- chan_iax2: Add ANI2/OLI information
element
Category: Channels/chan_pjsip
ASTERISK-29472: res_pjsip: OLI/ANI2 support missing
Reported by: N A
* [f8bf5e7b47] Naveen Albert -- res_pjsip_caller_id: Add ANI2/OLI
parsing
ASTERISK-29459: Missing configuration from PJSIP to SIP conversion script
Reported by: N A
* [c8bf8a54c2] Naveen Albert -- sip_to_pjsip: Fix missing cases
ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
failure SIP code
Reported by: Dan Cropp
* [55891227e8] Dan Cropp -- chan_pjsip, app_transfer: Add
TRANSFERSTATUSPROTOCOL variable
ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
* [134d2e729d] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
Category: Channels/chan_sip/General
ASTERISK-29083: Do not build chan_sip by default as it is now deprecated
Reported by: Sean Bright
* [52ca2323aa] Sean Bright -- chan_sip.c: Don't build by default
Category: Contrib/General
ASTERISK-29216: contrib: systemd asterisk service for centos8 or other
newer linux versions
Reported by: Mark Petersen
* [2d3441772b] Jaco Kroon -- contrib/systemd: Added note on common
issues with systemd and asterisk
Category: Core/Bridging
ASTERISK-29612: bridge_basic: Don't throw warning if attended transfer is
cancelled
Reported by: N A
* [4301fe20d1] Naveen Albert -- bridge_basic: Change warning to verbose
if transfer cancelled
Category: Core/General
ASTERISK-29544: Media Cache - Delayed remote sound file retrieve delays
all playbacks
Reported by: Andre Barbosa
* [2451dfd89f] Andre Barbosa -- media_cache: Don't lock when curl the
remote file
ASTERISK-29339: loader: Let's output warnings for deprecated modules!
Reported by: Joshua C. Colp
* [46ed6af9c2] Joshua C. Colp -- loader: Output warnings for deprecated
modules.
ASTERISK-29337: menuselect: Add ability to set deprecated in and removed
in versions for modules
Reported by: Joshua C. Colp
* [efc61a96f0] Joshua C. Colp -- menuselect: Add ability to set
deprecated and removed versions.
* [3330fb41f4] Joshua C. Colp -- xml: Allow deprecated_in and removed_in
for MODULEINFO.
ASTERISK-29335: xml: Embed module information into core XML documentation.
Reported by: Joshua C. Colp
* [149e5e5b86] Joshua C. Colp -- xml: Embed module information into core
XML documentation.
ASTERISK-29326: asterisk: Update copyright/company
Reported by: Joshua C. Colp
* [f8d1758792] Joshua C. Colp -- asterisk: Update copyright.
Category: Core/Logging
ASTERISK-29529: Add custom logging level
Reported by: N A
* [eb874f92db] Naveen Albert -- logger: Add custom logging capabilities
ASTERISK-29054: Logger: Add debug logging categories
Reported by: Kevin Harwell
* [56028426de] Kevin Harwell -- Logging: Add debug logging categories
Category: Core/Sorcery
ASTERISK-29321: sorcery: Add support for more intelligent reloading.
Reported by: Joshua C. Colp
* [304f8ddfb2] Joshua C. Colp -- sorcery: Add support for more
intelligent reloading.
Category: Documentation
ASTERISK-29335: xml: Embed module information into core XML documentation.
Reported by: Joshua C. Colp
* [149e5e5b86] Joshua C. Colp -- xml: Embed module information into core
XML documentation.
ASTERISK-29336: documentation: Fix inconsistent support levels
Reported by: Joshua C. Colp
* [7438586d8e] Joshua C. Colp -- documentation: Fix non-matching module
support levels.
Category: Formats/format_wav
ASTERISK-29275: Support of MIME-type for wav16
Reported by: Boris P. Korzun
* [ff493d6f7d] Sean Bright -- res_http_media_cache.c: Compare unaltered
MIME types.
* [a96eb6de6c] Boris P. Korzun -- format_wav: Support of MIME-type for
wav16
Category: Functions/func_math
ASTERISK-29495: Return integer instead of float if response is a whole
number
Reported by: N A
* [d6034df64a] Naveen Albert -- func_math: Return integer instead of
float if possible
Category: Functions/func_vmcount
ASTERISK-29661: func_vmcount: Add support for multiple mailboxes
Reported by: N A
* [13ec117595] Naveen Albert -- func_vmcount: Add support for multiple
mailboxes
Category: Resources/General
ASTERISK-29056: Increase reg_server column size for ps_contacts table
realtime
Reported by: sungtae kim
* [9052e448ec] Sungtae Kim -- realtime: Increased reg_server character
size
Category: Resources/res_ari_playbacks
ASTERISK-29501: ARI - Stasis Playback doesn't hangup call when processing
a list of invalid files
Reported by: Andre Barbosa
* [f4d3f021f9] Andre Barbosa -- res_stasis_playback: Check for chan
hangup on play_on_channels
Category: Resources/res_http_media_cache
ASTERISK-29527: res_http_media_cache: Cleanup audio format lookup in HTTP
requests
Reported by: Sean Bright
* [382143e58e] Sean Bright -- res_http_media_cache: Cleanup audio format
lookup in HTTP requests
ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in
/tmp
Reported by: laszlovl
* [b08427134f] laszlovl -- Introduce astcachedir, to be used for
temporary bucket files
Category: Resources/res_musiconhold
ASTERISK-29262: Support of various URL-schemes by MoH
Reported by: Boris P. Korzun
* [92f5cf7f2d] Boris P. Korzun -- res_musiconhold: Add support of
various URL-schemes by MoH.
Category: Resources/res_pjsip_caller_id
ASTERISK-29472: res_pjsip: OLI/ANI2 support missing
Reported by: N A
* [f8bf5e7b47] Naveen Albert -- res_pjsip_caller_id: Add ANI2/OLI
parsing
Category: Resources/res_pjsip_dtmf_info
ASTERISK-29460: Recognize application/hook-flash in PJSIP
Reported by: N A
* [1b38e89734] Naveen Albert -- res_pjsip_dtmf_info: Hook flash
Category: Resources/res_pjsip_registrar
ASTERISK-29325: res_pjsip_registrar: Include source IP address and port in
log messages
Reported by: Joshua C. Colp
* [6f67f24afd] Joshua C. Colp -- res_pjsip_registrar: Include source IP
and port in log messages.
Category: Resources/res_pjsip_session
ASTERISK-28549: Two repeated 183
Reported by: Gant Liu
* [134d2e729d] Ivan Poddubnyi -- chan_pjsip: Stop queueing control
frames twice on outgoing channels
Category: Resources/res_rtp_asterisk
ASTERISK-29508: STUN server address refresh
Reported by: Sébastien Duthil
* [18189ff594] Sebastien Duthil -- res_rtp_asterisk: Automatically
refresh stunaddr from DNS
ASTERISK-29434: Asterisk reveals pjproject version in STUN packets
Reported by: Jeremy Lainé
* [d162789c4d] Jeremy Lainé -- res_rtp_asterisk: make it possible to
remove SOFTWARE attribute
Category: Resources/res_stasis
ASTERISK-29055: Create a Bridge with video_single mode
Reported by: sungtae kim
* [aae0904c7d] Sungtae Kim -- res_stasis.c: Added video_single option
for bridge creation
Category: Resources/res_stasis_playback
ASTERISK-29464: ARI - PlaybackFinish skip error events
Reported by: Andre Barbosa
* [a47308ccb2] Andre Barbosa -- res_stasis_playback: Send PlaybackFinish
event only once for errors
Category: pjproject/pjsip
ASTERISK-29525: PJSIP remove_existing unavailable contacts
Reported by: Joseph Nadiv
* [6a04c43035] Joseph Nadiv -- res_pjsip_registrar: Remove unavailable
contacts if exceeds max_contacts
----------------------------------------------------------------------
Open Issues
[Back to Top]
This is a list of all open issues from the issue tracker that were
referenced by changes that went into this release.
Bug
Category: Applications/app_voicemail/ODBC
ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
Reported by: Schneur Rosenberg
* [44d68bd56b] Sean Bright -- app_voicemail: Prevent deadlocks when out
of ODBC database connections
Category: Resources/res_pjsip_endpoint_identifier_ip
ASTERISK-29624: Contact identifier is not updated when FDQN resolves to a
new address
Reported by: Philip Young
* [91b0778791] George Joseph -- chan_iax2.c: Require secret and auth
method if encryption is enabled
Category: Resources/res_pjsip_outbound_registration
ASTERISK-29241: pjsip / register: wrong port used in Contact and Via if
multiple transports are defined.
Reported by: Michael Maier
* [f160725fc4] Bernd Zobl -- res_pjsip/pjsip_message_filter: set
preferred transport in pjsip_message_filter
Category: Resources/res_srtp
ASTERISK-29625: srtp cryptos accepted if not enabled
Reported by: Jasper Hafkenscheid
* [f1e1f9f37f] Jasper Hafkenscheid -- res_srtp: Disable parsing of not
enabled cryptos
----------------------------------------------------------------------
Commits Not Associated with an Issue
[Back to Top]
This is a list of all changes that went into this release that did not
reference a JIRA issue.
+------------------------------------------------------------------------+
| Revision | Author | Summary |
|------------+---------------+-------------------------------------------|
| | Asterisk | |
| 9ff955f4d1 | Development | Update CHANGES and UPGRADE.txt for 19.0.0 |
| | Team | |
|------------+---------------+-------------------------------------------|
| 9175012a12 | Sean Bright | Makefile: Use basename in a |
| | | POSIX-compliant way. |
|------------+---------------+-------------------------------------------|
| 1f5ac24fa3 | Mark Murawski | pbx_ael: Fix crash and lockup issue |
| | | regarding 'ael reload' |
|------------+---------------+-------------------------------------------|
| 245778a756 | Sean Bright | app_externalivr.c: Fix mixed leading |
| | | whitespace in source code. |
|------------+---------------+-------------------------------------------|
| f26505d615 | Sean Bright | test_http_media_cache.c: Fix copy/paste |
| | | error during test deregistration. |
|------------+---------------+-------------------------------------------|
| f22b413ece | Alexander | dialplan: Add one static and fix two |
| | Traud | whitespace errors. |
|------------+---------------+-------------------------------------------|
| 73e2288db7 | Alexander | BuildSystem: Remove two dead exceptions |
| | Traud | for compiler Clang. |
|------------+---------------+-------------------------------------------|
| 90c9c90b11 | Joshua C. | docs: Remove embedded macro in |
| | Colp | WaitForCond XML documentation. |
|------------+---------------+-------------------------------------------|
| 0ac346ec47 | Ben Ford | Update default branch for Asterisk 19. |
|------------+---------------+-------------------------------------------|
| 237285a9a8 | Sean Bright | res_pjsip_stir_shaken: RFC 8225 |
| | | compliance and error message cleanup. |
|------------+---------------+-------------------------------------------|
| 785e4afc20 | Sean Bright | main/cdr.c: Correct Party A selection. |
|------------+---------------+-------------------------------------------|
| b7027de195 | George Joseph | res_pjsip_messaging: Overwrite user in |
| | | existing contact URI |
|------------+---------------+-------------------------------------------|
| 56c2cc474b | Jaco Kroon | func_lock: Add "dialplan locks show" cli |
| | | command. |
|------------+---------------+-------------------------------------------|
| 19a8383a1f | Jaco Kroon | func_lock: Prevent module unloading |
| | | in-use module. |
|------------+---------------+-------------------------------------------|
| e8875d5ca1 | Jaco Kroon | func_lock: Fix memory corruption during |
| | | unload. |
|------------+---------------+-------------------------------------------|
| caceba7988 | Jaco Kroon | func_lock: Fix requesters counter in |
| | | error paths. |
|------------+---------------+-------------------------------------------|
| c0fc8adbb6 | Sean Bright | menuselect: Fix description of several |
| | | modules. |
|------------+---------------+-------------------------------------------|
| 12e8600849 | Ben Ford | STIR/SHAKEN: Add Date header, dest->tn, |
| | | and URL checking. |
|------------+---------------+-------------------------------------------|
| 987f5eb0ad | Joshua C. | asterisk: We've moved to Libera Chat! |
| | Colp | |
|------------+---------------+-------------------------------------------|
| 0564d12280 | Ben Ford | STIR/SHAKEN: Switch to base64 URL |
| | | encoding. |
|------------+---------------+-------------------------------------------|
| 05f7bc9c66 | Ben Ford | STIR/SHAKEN: OPENSSL_free serial hex from |
| | | openssl. |
|------------+---------------+-------------------------------------------|
| 259ecfa289 | Ben Ford | STIR/SHAKEN: Fix certificate type and |
| | | storage. |
|------------+---------------+-------------------------------------------|
| 09303e8e22 | George Joseph | Updates for the MessageSend Dialplan App |
|------------+---------------+-------------------------------------------|
| e39efabd97 | Sean Bright | translate.c: Avoid refleak when checking |
| | | for a translation path |
|------------+---------------+-------------------------------------------|
| 531eb65cf3 | Joshua C. | svn: Switch to https scheme. |
| | Colp | |
|------------+---------------+-------------------------------------------|
| 512d38868c | George Joseph | res_pjsip: Update documentation for the |
| | | auth object |
|------------+---------------+-------------------------------------------|
| 45a1977de4 | Ben Ford | res_aeap: Add basic config skeleton and |
| | | CLI commands. |
|------------+---------------+-------------------------------------------|
| 5a13e95c56 | Sean Bright | loader.c: Speed up deprecation metadata |
| | | lookup |
|------------+---------------+-------------------------------------------|
| c4a376aac2 | Kevin Harwell | res_rtp_asterisk: Don't count 0 as a |
| | | minimum lost packets |
|------------+---------------+-------------------------------------------|
| 65b68fd060 | Kevin Harwell | res_rtp_asterisk: Statically declare |
| | | rtp_drop_packets_data object |
|------------+---------------+-------------------------------------------|
| b86f1ef54c | Kevin Harwell | res_rtp_asterisk: Add a DEVMODE RTP drop |
| | | packets CLI command |
|------------+---------------+-------------------------------------------|
| 623abc2b6a | Joshua C. | res_pjsip: Give error when TLS transport |
| | Colp | configured but not supported. |
|------------+---------------+-------------------------------------------|
| eb92fb7298 | Kevin Harwell | time: Add timeval create and unit |
| | | conversion functions |
|------------+---------------+-------------------------------------------|
| 25758670b8 | Ben Ford | logger.conf.sample: Add more debug |
| | | documentation. |
|------------+---------------+-------------------------------------------|
| 55c53de022 | Ben Ford | logging: Add .log to samples and update |
| | | asterisk.logrotate. |
|------------+---------------+-------------------------------------------|
| 41389bfdbd | Jaco Kroon | func_callerid+res_agi: Fix compile errors |
| | | related to -Werror=zero-length-bounds |
|------------+---------------+-------------------------------------------|
| 8acb4fbd1e | Jaco Kroon | app.h: Fix -Werror=zero-length-bounds |
| | | compile errors in dev mode. |
|------------+---------------+-------------------------------------------|
| 1ae40e502d | Alexander | res_format_attr_*: Parameter Names are |
| | Traud | Case-Insensitive. |
|------------+---------------+-------------------------------------------|
| 8c461845c8 | Alexander | chan_iax2: System Header strings is |
| | Traud | included via asterisk.h/compat.h. |
|------------+---------------+-------------------------------------------|
| df37b8181c | Sean Bright | res_musiconhold.c: Plug ref leak caused |
| | | by ao2_replace() misuse. |
|------------+---------------+-------------------------------------------|
| 607603cf89 | George Joseph | res_pjsip_refer: Move the progress dlg |
| | | release to a serializer |
|------------+---------------+-------------------------------------------|
| a34e7de61c | Alexander | res_format_attr_h263: Generate valid SDP |
| | Traud | fmtp for H.263+. |
|------------+---------------+-------------------------------------------|
| e5e49d7ecd | Kevin Harwell | res_rtp_asterisk: Add packet subtype |
| | | during RTCP debug when relevant |
|------------+---------------+-------------------------------------------|
| 5894535fed | Alexander | chan_sip: Filter pass-through audio/video |
| | Traud | formats away, again. |
|------------+---------------+-------------------------------------------|
| b0f349a330 | Jaco Kroon | func_odbc: Introduce minargs config and |
| | | expose ARGC in addition to ARGn. |
|------------+---------------+-------------------------------------------|
| 15b4080679 | George Joseph | res_pjsip_refer: Always serialize calls |
| | | to refer_progress_notify |
|------------+---------------+-------------------------------------------|
| 4a71b08091 | Sean Bright | app_read: Release tone zone reference on |
| | | early return. |
|------------+---------------+-------------------------------------------|
| 05472da92b | Ivan | main/frame: Add missing control frame |
| | Poddubnyi | names to ast_frame_subclass2str |
|------------+---------------+-------------------------------------------|
| 060ce10163 | Jaco Kroon | AC_HEADER_STDC causes a compile failure |
| | | with autoconf 2.70 |
|------------+---------------+-------------------------------------------|
| 10a0a0c59b | Alexander | pjsip_scheduler: Fix pjsip show |
| | Traud | scheduled_tasks like for compiler Clang. |
|------------+---------------+-------------------------------------------|
| 6d2bec7028 | Sean Bright | res_pjsip_pubsub: Fix truncation of |
| | | persisted SUBSCRIBE packet |
|------------+---------------+-------------------------------------------|
| 948ceb1228 | Ben Ford | chan_pjsip.c: Add parameters to frame in |
| | | indicate. |
|------------+---------------+-------------------------------------------|
| 4e038c1eaa | Jaco Kroon | pbx_lua: Add LUA_VERSIONS environment |
| | | variable to ./configure. |
|------------+---------------+-------------------------------------------|
| 1b74555fcf | Sean Bright | asterisk: Export additional manager |
| | | functions |
|------------+---------------+-------------------------------------------|
| 80c14f74bc | Alexander | codecs: Remove test-law. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| 058bc0d593 | Richard | chan_vpb.cc: Fix compile errors. |
| | Mudgett | |
|------------+---------------+-------------------------------------------|
| 6d7af72559 | Richard | res_pjsip_session.c: Fix compiler |
| | Mudgett | warnings. |
|------------+---------------+-------------------------------------------|
| 9ee1f7154f | Joshua C. | res_pjsip_pidf_digium_body_supplement: |
| | Colp | Support Sangoma user agent. |
|------------+---------------+-------------------------------------------|
| c8b6340023 | Sean Bright | media_cache: Fix reference leak with |
| | | bucket file metadata |
|------------+---------------+-------------------------------------------|
| d04b5903d1 | Sean Bright | CHANGES: Remove already applied CHANGES |
| | | update |
|------------+---------------+-------------------------------------------|
| 7c355d78cb | Alexander | modules.conf: Align the comments for more |
| | Traud | conclusiveness. |
|------------+---------------+-------------------------------------------|
| 2fe76dd816 | George Joseph | res_pjsip_outbound_registration.c: Use |
| | | our own scheduler and other stuff |
|------------+---------------+-------------------------------------------|
| 5a4640d208 | George Joseph | pjsip_scheduler.c: Add type ONESHOT and |
| | | enhance cli show command |
|------------+---------------+-------------------------------------------|
| cc7eb72f65 | Alexei | sched: AST_SCHED_REPLACE_UNREF can lead |
| | Gradinari | to use after free of data |
|------------+---------------+-------------------------------------------|
| 64d2de19ee | Alexander | res_stir_shaken: Include OpenSSL headers |
| | Traud | where used actually. |
|------------+---------------+-------------------------------------------|
| cd32317691 | Alexander | chan_sip: On authentication, pick MD5 for |
| | Traud | sure. |
|------------+---------------+-------------------------------------------|
| 1650d50e91 | Walter Doekes | main/say: Work around gcc 9 |
| | | format-truncation false positive |
|------------+---------------+-------------------------------------------|
| c62193c5de | Kevin Harwell | res_pjsip, res_pjsip_session: initialize |
| | | local variables |
|------------+---------------+-------------------------------------------|
| f3452c85e5 | Alexander | install_prereq: Add GMime 3.0. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| db4320a6a0 | Alexander | BuildSystem: Enable Lua 5.4. |
| | Traud | |
|------------+---------------+-------------------------------------------|
| 773f424c7f | George Joseph | app_confbridge/bridge_softmix: Add |
| | | ability to force estimated bitrate |
|------------+---------------+-------------------------------------------|
| e7bd97e2e5 | Torrey Searle | res_pjsip_diversion: fix double 181 |
|------------+---------------+-------------------------------------------|
| 505211551a | Sean Bright | res_musiconhold: Clarify that playlist |
| | | mode only supports HTTP(S) URLs |
|------------+---------------+-------------------------------------------|
| 23e427bbd2 | Joshua C. | res_pjsip_session: Fix stream name memory |
| | Colp | leak. |
|------------+---------------+-------------------------------------------|
| 923d95cc84 | George Joseph | logger.h: Fix ast_trace to respect |
| | | scope_level |
|------------+---------------+-------------------------------------------|
| 5a0e1d256d | Sean Bright | audiosocket: Fix module menuselect |
| | | descriptions |
|------------+---------------+-------------------------------------------|
| 39bb45cdfc | George Joseph | bridge_softmix/sfu_topologies_on_join: |
| | | Ignore topology change failures |
|------------+---------------+-------------------------------------------|
| bc038e6191 | Sean Bright | res_pjsip_session.c: Fix build when |
| | | TEST_FRAMEWORK is not defined |
|------------+---------------+-------------------------------------------|
| 44bb0858cb | George Joseph | debugging: Add enough to choke a mule |
|------------+---------------+-------------------------------------------|
| 80a609fcce | Ben Ford | Bridging: Use a ref to bridge_channel's |
| | | channel to prevent crash. |
|------------+---------------+-------------------------------------------|
| | Patrick | res_pjsip_session: Deferred re-INVITE |
| f8fe20eb9f | Verzele | without SDP send a=sendrecv instead of |
| | | a=sendonly |
|------------+---------------+-------------------------------------------|
| 1a5597741f | Kevin Harwell | conversions: Add string to signed integer |
| | | conversion functions |
|------------+---------------+-------------------------------------------|
| 5989e0de0f | George Joseph | ast_coredumper: Fix issues with naming |
|------------+---------------+-------------------------------------------|
| f225e9bf35 | Alexander | sip_nat_settings: Update script for |
| | Traud | latest Linux. |
|------------+---------------+-------------------------------------------|
| 8907a9f0b9 | Alexander | samples: Fix keep_alive_interval default |
| | Traud | in pjsip.conf. |
|------------+---------------+-------------------------------------------|
| 54ddf19141 | George Joseph | logger.c: Added a new log formatter |
| | | called "plain" |
|------------+---------------+-------------------------------------------|
| 057fda460b | Sean Bright | res_musiconhold.c: Use ast_file_read_dir |
| | | to scan MoH directory |
|------------+---------------+-------------------------------------------|
| 64ca2d48da | George Joseph | scope_trace: Added debug messages and |
| | | added additional macros |
|------------+---------------+-------------------------------------------|
| 118cb3f0dd | George Joseph | stream.c: Added 2 more debugging utils |
| | | and added pos to stream string |
|------------+---------------+-------------------------------------------|
| 647c53c41f | George Joseph | ACN: Changes specific to the core |
|------------+---------------+-------------------------------------------|
| 447f6cc37a | Joshua C. | res_pjsip: Fix codec preference defaults. |
| | Colp | |
|------------+---------------+-------------------------------------------|
| 048b12b59d | Sean Bright | vector.h: Fix implementation of |
| | | AST_VECTOR_COMPACT() for empty vectors |
|------------+---------------+-------------------------------------------|
| 9ed6387c14 | Ben Ford | utils.c: NULL terminate |
| | | ast_base64decode_string. |
|------------+---------------+-------------------------------------------|
| a15e64aaf5 | George Joseph | ACN: Configuration renaming for pjsip |
| | | endpoint |
|------------+---------------+-------------------------------------------|
| deaa3742dc | Ben Ford | res_stir_shaken: Fix memory allocation |
| | | error in curl.c |
|------------+---------------+-------------------------------------------|
| 1f78ee9d0f | George Joseph | res_pjsip_session: Ensure reused streams |
| | | have correct bundle group |
|------------+---------------+-------------------------------------------|
| 7d96b3e437 | Sean Bright | utf8.c: Add UTF-8 validation and utility |
| | | functions |
|------------+---------------+-------------------------------------------|
| b5bb4a7a0d | Sean Bright | vector.h: Add AST_VECTOR_SORT() |
|------------+---------------+-------------------------------------------|
| e1d30f3e6c | George Joseph | CI: Force publishAsteriskDocs to use |
| | | python2 |
|------------+---------------+-------------------------------------------|
| 9f641483e6 | Joshua C. | websocket / pjsip: Increase maximum |
| | Colp | packet size. |
|------------+---------------+-------------------------------------------|
| 9c3b57822a | George Joseph | Prepare master for the next Asterisk |
| | | version |
|------------+---------------+-------------------------------------------|
| f1d7de121f | Joshua C. | pjsip: Include timer patch to prevent |
| | Colp | cancelling timer 0. |
+------------------------------------------------------------------------+
----------------------------------------------------------------------
Diffstat Results
[Back to Top]
This is a summary of the changes to the source code that went into this
release that was generated using the diffstat utility.
.lastclean | 1
.version | 1
ChangeLog |92149 ----------
addons/app_mysql.c | 667
addons/cdr_mysql.c | 758
apps/app_dahdiras.c | 231
apps/app_fax.c | 1003
apps/app_ices.c | 214
apps/app_image.c | 107
apps/app_nbscat.c | 223
apps/app_url.c | 180
asterisk-18.0.0-summary.html | 1162
asterisk-18.0.0-summary.txt | 2873
b/.gitreview | 2
b/CHANGES | 416
b/Makefile | 14
b/README.md | 8
b/UPGRADE.txt | 224
b/addons/Makefile | 4
b/addons/ooh323c/src/ooq931.c | 2
b/apps/app_agent_pool.c | 10
b/apps/app_attended_transfer.c | 2
b/apps/app_blind_transfer.c | 2
b/apps/app_chanspy.c | 6
b/apps/app_confbridge.c | 93
b/apps/app_dial.c | 93
b/apps/app_directory.c | 2
b/apps/app_dtmfstore.c | 286
b/apps/app_externalivr.c | 288
b/apps/app_macro.c | 2
b/apps/app_meetme.c | 6
b/apps/app_mf.c | 361
b/apps/app_milliwatt.c | 23
b/apps/app_mixmonitor.c | 98
b/apps/app_morsecode.c | 168
b/apps/app_mp3.c | 24
b/apps/app_originate.c | 122
b/apps/app_osplookup.c | 7
b/apps/app_page.c | 13
b/apps/app_queue.c | 345
b/apps/app_read.c | 36
b/apps/app_reload.c | 110
b/apps/app_speech_utils.c | 2
b/apps/app_stack.c | 4
b/apps/app_talkdetect.c | 2
b/apps/app_transfer.c | 24
b/apps/app_verbose.c | 9
b/apps/app_voicemail.c | 81
b/apps/app_waitforcond.c | 234
b/apps/confbridge/conf_config_parser.c | 34
b/apps/confbridge/include/confbridge.h | 3
b/bridges/bridge_softmix.c | 154
b/build_tools/install_subst | 1
b/build_tools/make_defaults_h | 1
b/build_tools/menuselect-deps.in | 8
b/build_tools/mkpkgconfig | 1
b/cdr/cdr_adaptive_odbc.c | 2
b/channels/Makefile | 5
b/channels/chan_alsa.c | 8
b/channels/chan_audiosocket.c | 5
b/channels/chan_dahdi.c | 18
b/channels/chan_dahdi.h | 16
b/channels/chan_iax2.c | 103
b/channels/chan_mgcp.c | 42
b/channels/chan_pjsip.c | 341
b/channels/chan_sip.c | 128
b/channels/chan_skinny.c | 7
b/channels/iax2/codec_pref.c | 2
b/channels/iax2/format_compatibility.c | 1
b/channels/iax2/include/iax2.h | 2
b/channels/iax2/include/parser.h | 1
b/channels/iax2/parser.c | 10
b/channels/sig_analog.c | 60
b/channels/sig_analog.h | 4
b/channels/sip/include/sip.h | 2
b/codecs/codec_dahdi.c | 2
b/codecs/codec_ulaw.c | 42
b/configs/basic-pbx/modules.conf | 8
b/configs/samples/aeap.conf.sample | 15
b/configs/samples/asterisk.conf.sample | 1
b/configs/samples/chan_dahdi.conf.sample | 18
b/configs/samples/confbridge.conf.sample | 9
b/configs/samples/features.conf.sample | 4
b/configs/samples/func_odbc.conf.sample | 11
b/configs/samples/iax.conf.sample | 9
b/configs/samples/logger.conf.sample | 33
b/configs/samples/modules.conf.sample | 39
b/configs/samples/musiconhold.conf.sample | 4
b/configs/samples/pjproject.conf.sample | 5
b/configs/samples/pjsip.conf.sample | 86
b/configs/samples/queues.conf.sample | 19
b/configs/samples/res_curl.conf.sample | 1
b/configs/samples/rtp.conf.sample | 20
b/configs/samples/stasis.conf.sample | 3
b/configs/samples/statsd.conf.sample | 3
b/configs/samples/stir_shaken.conf.sample | 44
b/configure | 1450
b/configure.ac | 98
b/contrib/ast-db-manage/config/versions/1ae0609b6646_increse_reg_server_size.py | 22
b/contrib/ast-db-manage/config/versions/8915fcc5766f_add_ringinuse_to_queue_members.py | 30
b/contrib/ast-db-manage/config/versions/a06d8f8462d9_add_t38_bind_udptl_to_media_address.py | 29
b/contrib/ast-db-manage/config/versions/c20d6e3992f4_add_allow_unauthenticated_options.py | 29
b/contrib/ast-db-manage/config/versions/e658c26033ca_create_history_info_flag.py | 38
b/contrib/ast-db-manage/config/versions/f56d79a9f337_pjsip_create_remove_unavailable.py | 30
b/contrib/scripts/asterisk.logrotate | 2
b/contrib/scripts/get_mp3_source.sh | 2
b/contrib/scripts/install_prereq | 12
b/contrib/scripts/sip_to_pjsip/astconfigparser.py | 43
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 8
b/contrib/systemd/asterisk.service | 7
b/doc/appdocsxml.dtd | 26
b/formats/format_ogg_speex.c | 9
b/formats/format_wav.c | 3
b/funcs/func_callerid.c | 146
b/funcs/func_channel.c | 38
b/funcs/func_curl.c | 48
b/funcs/func_env.c | 87
b/funcs/func_frame_drop.c | 291
b/funcs/func_lock.c | 228
b/funcs/func_math.c | 185
b/funcs/func_odbc.c | 34
b/funcs/func_periodic_hook.c | 3
b/funcs/func_pjsip_aor.c | 2
b/funcs/func_pjsip_contact.c | 2
b/funcs/func_pjsip_endpoint.c | 2
b/funcs/func_sayfiles.c | 396
b/funcs/func_scramble.c | 235
b/funcs/func_strings.c | 144
b/funcs/func_vmcount.c | 23
b/funcs/func_volume.c | 48
b/include/asterisk/app.h | 24
b/include/asterisk/autoconfig.h.in | 31
b/include/asterisk/bridge.h | 14
b/include/asterisk/bridge_channel.h | 14
b/include/asterisk/channel.h | 23
b/include/asterisk/core_unreal.h | 2
b/include/asterisk/doxygen/licensing.h | 3
b/include/asterisk/dsp.h | 4
b/include/asterisk/format_cache.h | 18
b/include/asterisk/format_compatibility.h | 2
b/include/asterisk/logger.h | 17
b/include/asterisk/logger_category.h | 178
b/include/asterisk/manager.h | 6
b/include/asterisk/paths.h | 1
b/include/asterisk/pbx.h | 8
b/include/asterisk/res_pjsip.h | 151
b/include/asterisk/res_pjsip_session.h | 8
b/include/asterisk/res_stir_shaken.h | 11
b/include/asterisk/rtp_engine.h | 79
b/include/asterisk/say.h | 100
b/include/asterisk/sched.h | 5
b/include/asterisk/sorcery.h | 22
b/include/asterisk/stasis_app_playback.h | 2
b/include/asterisk/stasis_channels.h | 33
b/include/asterisk/statsd.h | 6
b/include/asterisk/stream.h | 4
b/include/asterisk/strings.h | 4
b/include/asterisk/stun.h | 25
b/include/asterisk/time.h | 79
b/include/asterisk/utils.h | 60
b/main/abstract_jb.c | 26
b/main/app.c | 21
b/main/asterisk.c | 16
b/main/bridge.c | 44
b/main/bridge_basic.c | 9
b/main/bridge_channel.c | 32
b/main/bucket.c | 3
b/main/cdr.c | 2
b/main/channel.c | 95
b/main/channel_internal_api.c | 2
b/main/cli.c | 51
b/main/codec_builtin.c | 16
b/main/config_options.c | 60
b/main/core_local.c | 3
b/main/core_unreal.c | 92
b/main/dns.c | 17
b/main/dns_recurring.c | 9
b/main/dsp.c | 45
b/main/file.c | 1
b/main/fixedjitterbuf.c | 2
b/main/format_cache.c | 29
b/main/format_cap.c | 2
b/main/format_compatibility.c | 7
b/main/frame.c | 9
b/main/indications.c | 6
b/main/loader.c | 183
b/main/logger.c | 214
b/main/logger_category.c | 324
b/main/manager.c | 6
b/main/manager_channels.c | 95
b/main/media_cache.c | 89
b/main/message.c | 100
b/main/options.c | 7
b/main/pbx.c | 14
b/main/pbx_builtins.c | 137
b/main/pbx_include.c | 2
b/main/pbx_timing.c | 2
b/main/pbx_variables.c | 2
b/main/rtp_engine.c | 68
b/main/say.c | 558
b/main/sorcery.c | 17
b/main/stasis.c | 4
b/main/stasis_channels.c | 12
b/main/stream.c | 30
b/main/stun.c | 83
b/main/tcptls.c | 12
b/main/term.c | 105
b/main/time.c | 145
b/main/translate.c | 32
b/main/utils.c | 129
b/makeopts.in | 21
b/menuselect/configure | 14
b/menuselect/menuselect.c | 36
b/menuselect/menuselect.h | 2
b/menuselect/menuselect_curses.c | 10
b/menuselect/menuselect_newt.c | 10
b/pbx/pbx_ael.c | 7
b/pbx/pbx_realtime.c | 32
b/res/Makefile | 5
b/res/ari/resource_bridges.c | 19
b/res/ari/resource_bridges.h | 4
b/res/ari/resource_channels.c | 32
b/res/ari/resource_endpoints.c | 1
b/res/parking/parking_bridge_features.c | 1
b/res/prometheus/bridges.c | 12
b/res/prometheus/channels.c | 15
b/res/prometheus/endpoints.c | 9
b/res/res_aeap.c | 298
b/res/res_agi.c | 6
b/res/res_audiosocket.c | 3
b/res/res_calendar.c | 8
b/res/res_config_pgsql.c | 32
b/res/res_fax.c | 14
b/res/res_format_attr_celt.c | 14
b/res/res_format_attr_h263.c | 141
b/res/res_format_attr_ilbc.c | 15
b/res/res_format_attr_opus.c | 31
b/res/res_format_attr_silk.c | 17
b/res/res_format_attr_siren14.c | 13
b/res/res_format_attr_siren7.c | 13
b/res/res_format_attr_vp8.c | 12
b/res/res_hep_pjsip.c | 2
b/res/res_http_media_cache.c | 117
b/res/res_http_websocket.c | 2
b/res/res_monitor.c | 3
b/res/res_musiconhold.c | 41
b/res/res_odbc.c | 1
b/res/res_odbc_transaction.c | 5
b/res/res_parking.c | 1
b/res/res_pjproject.c | 24
b/res/res_pjsip.c | 256
b/res/res_pjsip/config_transport.c | 47
b/res/res_pjsip/location.c | 1
b/res/res_pjsip/pjsip_configuration.c | 22
b/res/res_pjsip/pjsip_message_filter.c | 11
b/res/res_pjsip/pjsip_options.c | 2
b/res/res_pjsip/pjsip_scheduler.c | 180
b/res/res_pjsip/pjsip_transport_management.c | 2
b/res/res_pjsip_authenticator_digest.c | 27
b/res/res_pjsip_caller_id.c | 59
b/res/res_pjsip_config_wizard.c | 15
b/res/res_pjsip_dialog_info_body_generator.c | 119
b/res/res_pjsip_diversion.c | 347
b/res/res_pjsip_dlg_options.c | 2
b/res/res_pjsip_dtmf_info.c | 10
b/res/res_pjsip_endpoint_identifier_ip.c | 3
b/res/res_pjsip_header_funcs.c | 192
b/res/res_pjsip_messaging.c | 833
b/res/res_pjsip_nat.c | 34
b/res/res_pjsip_outbound_authenticator_digest.c | 508
b/res/res_pjsip_outbound_registration.c | 13
b/res/res_pjsip_path.c | 12
b/res/res_pjsip_pidf_digium_body_supplement.c | 8
b/res/res_pjsip_pubsub.c | 12
b/res/res_pjsip_refer.c | 163
b/res/res_pjsip_registrar.c | 151
b/res/res_pjsip_sdp_rtp.c | 108
b/res/res_pjsip_session.c | 2179
b/res/res_pjsip_stir_shaken.c | 111
b/res/res_pjsip_t38.c | 52
b/res/res_pjsip_transport_websocket.c | 2
b/res/res_pktccops.c | 40
b/res/res_prometheus.c | 4
b/res/res_remb_modifier.c | 2
b/res/res_rtp_asterisk.c | 1190
b/res/res_sorcery_config.c | 12
b/res/res_srtp.c | 37
b/res/res_stasis.c | 31
b/res/res_stasis_playback.c | 33
b/res/res_stasis_snoop.c | 12
b/res/res_statsd.c | 16
b/res/res_stir_shaken.c | 260
b/res/res_stir_shaken/certificate.c | 32
b/res/res_stir_shaken/certificate.h | 12
b/res/res_stir_shaken/curl.c | 103
b/res/res_stir_shaken/curl.h | 10
b/res/res_stir_shaken/stir_shaken.c | 87
b/res/res_stir_shaken/stir_shaken.h | 12
b/res/res_stir_shaken/store.c | 20
b/res/res_tonedetect.c | 671
b/res/res_xmpp.c | 5
b/res/stasis/messaging.c | 72
b/res/stasis/stasis_bridge.c | 2
b/rest-api-templates/make_ari_stubs.py | 2
b/rest-api/api-docs/bridges.json | 6
b/rest-api/api-docs/playbacks.json | 3
b/rest-api/resources.json | 2
b/tests/CI/buildAsterisk.sh | 6
b/tests/CI/installAsterisk.sh | 1
b/tests/test_abstract_jb.c | 37
b/tests/test_http_media_cache.c | 79
b/tests/test_res_rtp.c | 40
b/tests/test_time.c | 170
b/third-party/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch | 37
b/third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch | 33
b/third-party/pjproject/patches/0090-Skip-unsupported-digest-algorithm-2408.patch | 212
b/third-party/pjproject/patches/0100-fix-double-stun-free.patch | 82
b/third-party/pjproject/patches/0110-tls-parent-listener-destroyed.patch | 166
b/third-party/pjproject/patches/0111-ssl-premature-destroy.patch | 136
b/third-party/pjproject/patches/0120-pjmedia_sdp_attr_get_rtpmap-Strip-param-trailing-whi.patch | 32
b/utils/.gitignore | 2
b/utils/Makefile | 22
b/utils/extconf.c | 4
cdr/cdr_syslog.c | 296
channels/chan_misdn.c |12838 -
channels/chan_nbs.c | 273
channels/chan_oss.c | 1527
channels/chan_phone.c | 1517
channels/chan_vpb.cc | 2878
channels/misdn/Makefile | 17
channels/misdn/chan_misdn_config.h | 172
channels/misdn/ie.c | 1414
channels/misdn/isdn_lib.c | 4819
channels/misdn/isdn_lib.h | 833
channels/misdn/isdn_lib_intern.h | 159
channels/misdn/isdn_msg_parser.c | 1769
channels/misdn/portinfo.c | 205
channels/misdn_config.c | 1273
configs/samples/cdr_mysql.conf.sample | 62
configs/samples/cdr_syslog.conf.sample | 83
configs/samples/misdn.conf.sample | 537
configs/samples/oss.conf.sample | 152
configs/samples/phone.conf.sample | 51
configs/samples/res_config_sqlite.conf.sample | 11
configs/samples/vpb.conf.sample | 248
contrib/realtime/mysql/mysql_cdr.sql | 41
contrib/realtime/mysql/mysql_config.sql | 1294
contrib/realtime/mysql/mysql_voicemail.sql | 35
contrib/realtime/postgresql/postgresql_cdr.sql | 45
contrib/realtime/postgresql/postgresql_config.sql | 1406
contrib/realtime/postgresql/postgresql_voicemail.sql | 39
doc/CHANGES-staging/hide_messaging_ami_events | 11
res/res_config_sqlite.c | 1787
utils/conf2ael.c | 729
utils/muted.c | 744
355 files changed, 17548 insertions(+), 141656 deletions(-)
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