From 2c1bba3cbec008c8ce35c78a2c79f9f207ea58bc Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Mon, 19 Oct 2020 13:31:06 -0500 Subject: [PATCH] Update for 18.0.0 --- .version | 2 +- ChangeLog | 4 + asterisk-18.0.0-rc2-summary.html | 15 - asterisk-18.0.0-rc2-summary.txt | 83 - asterisk-18.0.0-summary.html | 1162 ++++++++++++ asterisk-18.0.0-summary.txt | 2873 ++++++++++++++++++++++++++++++ 6 files changed, 4040 insertions(+), 99 deletions(-) delete mode 100644 asterisk-18.0.0-rc2-summary.html delete mode 100644 asterisk-18.0.0-rc2-summary.txt create mode 100644 asterisk-18.0.0-summary.html create mode 100644 asterisk-18.0.0-summary.txt diff --git a/.version b/.version index aef1cadeee..03191c968c 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -18.0.0-rc2 \ No newline at end of file +18.0.0 \ No newline at end of file diff --git a/ChangeLog b/ChangeLog index 4af3d494ed..a1bc97454d 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,7 @@ +2020-10-19 18:31 +0000 Asterisk Development Team + + * asterisk 18.0.0 Released. + 2020-10-13 16:24 +0000 Asterisk Development Team * asterisk 18.0.0-rc2 Released. diff --git a/asterisk-18.0.0-rc2-summary.html b/asterisk-18.0.0-rc2-summary.html deleted file mode 100644 index c8ea4c4547..0000000000 --- a/asterisk-18.0.0-rc2-summary.html +++ /dev/null @@ -1,15 +0,0 @@ -Release Summary - asterisk-18.0.0-rc2

Release Summary

asterisk-18.0.0-rc2

Date: 2020-10-13

<asteriskteam@digium.com>

Table of Contents

    -
  1. Summary
    2. -
  2. Contributors
    3. -
  3. Closed Issues
    4. -
  4. Diffstat
    5. -

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-rc1.

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.

- - -
CodersTestersReporters
1 Joshua C. Colp
1 Ross Beer

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.

Bug

Category: Resources/res_pjsip_session

ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
Reported by: Ross Beer
    -
  • [5cc4a391b3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call pref.
    • -

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.

configs/samples/pjsip.conf.sample          |   66 +++--------------------------
-res/res_pjsip.c                            |   20 ++++----
-res/res_pjsip/pjsip_configuration.c        |    2
-res/res_pjsip_session/pjsip_session_caps.c |   18 +++++--
-4 files changed, 32 insertions(+), 74 deletions(-)

\ No newline at end of file diff --git a/asterisk-18.0.0-rc2-summary.txt b/asterisk-18.0.0-rc2-summary.txt deleted file mode 100644 index 4501cfa2ad..0000000000 --- a/asterisk-18.0.0-rc2-summary.txt +++ /dev/null @@ -1,83 +0,0 @@ - Release Summary - - asterisk-18.0.0-rc2 - - Date: 2020-10-13 - - - - ---------------------------------------------------------------------- - - Table of Contents - - 1. Summary - 2. Contributors - 3. Closed Issues - 4. 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-rc1. - - ---------------------------------------------------------------------- - - 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 - 1 Joshua C. Colp 1 Ross Beer - - ---------------------------------------------------------------------- - - 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. - - Bug - - Category: Resources/res_pjsip_session - - ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due - to codec negotiation after upgrading from Asterisk 16 - Reported by: Ross Beer - * [5cc4a391b3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call - pref. - - ---------------------------------------------------------------------- - - 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. - - configs/samples/pjsip.conf.sample | 66 +++-------------------------- - res/res_pjsip.c | 20 ++++---- - res/res_pjsip/pjsip_configuration.c | 2 - res/res_pjsip_session/pjsip_session_caps.c | 18 +++++-- - 4 files changed, 32 insertions(+), 74 deletions(-) diff --git a/asterisk-18.0.0-summary.html b/asterisk-18.0.0-summary.html new file mode 100644 index 0000000000..99daa66d6c --- /dev/null +++ b/asterisk-18.0.0-summary.html @@ -0,0 +1,1162 @@ +Release Summary - asterisk-18.0.0

Release Summary

asterisk-18.0.0

Date: 2020-10-19

<asteriskteam@digium.com>

Table of Contents

    +
  1. Summary
    2. +
  2. Contributors
    3. +
  3. Closed Issues
    4. +
  4. Open Issues
    5. +
  5. Other Changes
    6. +
  6. Diffstat
    7. +

Summary

[Back to Top]

This is the first release of a major new version of Asterisk. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is a new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.

The data in this summary reflects changes that have been made since the previous release, asterisk-17.0.0.

Contributors

[Back to Top]

This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.

+ + +
CodersTestersReporters
70 Sean Bright
54 George Joseph
47 Joshua C. Colp
31 Kevin Harwell
26 Alexander Traud
17 Ben Ford
16 Joshua Colp
16 Jaco Kroon
11 Walter Doekes
9 sungtae kim
9 Alexander Traud
8 Torrey Searle
8 Corey Farrell
7 Richard Mudgett
6 Frederic LE FOLL
6 Pirmin Walthert
6 Guido Falsi
5 Asterisk Development Team
5 Alexei Gradinari
3 Pascal Cadotte Michaud
3 Igor Goncharovsky
3 Jean Aunis
3 Nickolay Shmyrev
3 lvl
2 Rodrigo Ramírez Norambuena
2 Andrew Siplas
2 Salah Ahmed
2 Michael Neuhauser
2 cmaj
1 Kevin Reeves
1 Michael Goryainov
1 Università di Bologna - CESIA VoIP
1 Chris Savinovich
1 Nathan Bruning
1 Paulo Vicentini
1 Peter Turczak
1 Sungtae Kim
1 Dan Cropp
1 Jared Smith
1 Stas Kobzar
1 Daniel Heckl
1 Dennis Buteyn
1 Bernard Merindol
1 Jonathan Rose
1 Kfir Itzhak
1 Roger James
1 Sebastian Kemper
1 Christoph Moench-Tegeder
1 Boris P. Korzun
1 Evandro César Arruda
1 Moises Silva
1 Chris-Savinovich
1 Michael Cargile
1 Sylvain Afchain
1 Florian Floimair
1 Nicholas John Koch
1 Peter Sokolov (License #7070)
1 Martin Tomec
1 Thomas Arimont (license 5525)
1 Seán C McCord
1 Patrick Verzele
1 snuffy
1 Sebastien Duthil
1 Jason Hord (license 6978)
1 tests/test_utils.c.
26 Joshua C. Colp
23 Alexander Traud
13 Kevin Harwell
9 Ross Beer
8 sungtae kim
8 nappsoft
7 Walter Doekes
7 Torrey Searle
6 Frederic LE FOLL
5 Guido Falsi
5 cmaj
5 George Joseph
4 Jaco Kroon
4 Jean Aunis - Prescom
4 Pascal Cadotte Michaud
4 Ross Beer
3 Joshua Elson
3 Sean Bright
3 Salah Ahmed
3 Nickolay V. Shmyrev
3 lvl
3 Alexei Gradinari
2 Stas Kobzar
2 Michael Neuhauser
2 Ruddy G
2 Joeran Vinzens
2 Timothy Vanderaerden
2 Sébastien Duthil
2 Peter Sokolov
2 Joseph Ades
2 Gregory Massel
2 Andrew Siplas
2 Jared Smith
2 Jonathan Harris
2 Michael Neuhauser
1 Ramarajan
1 Andrey V. T.
1 tootai
1 Martin Tomec
1 AvayaXAsterisk
1 Joshua C. Colp
1 Etienne Lessard
1 Benjamin Keith Ford
1 Yoooooo Ha
1 kevin@phoneburner.com
1 Juan Martin
1 Sylvain Afchain
1 Speed Dial Dave
1 Andrew Yager
1 Jean-Denis Girard
1 Marian Piater
1 Bernard Merindol
1 Martin Zeh
1 Corey Farrell
1 Dan Cropp
1 Moises Silva
1 Alexey Vasilyev
1 Thomas Johnson
1 Seán C. McCord
1 Dirk Wendland
1 Bryan Nelson
1 Sam Banks
1 Misha Vodsedalek
1 Nicholas John Koch
1 Richard Kenner
1 EDV O-TON
1 Byron Clark
1 Christoph Moench-Tegeder
1 sstream
1 Dmitriy Serov
1 Alex
1 candrews
1 Sébastien Duthil
1 Robert Sutton
1 Evandro César Arruda
1 Paul Brooks
1 Yury Kirsanov
1 Jason Hord
1 Michael Cargile
1 Kevin Flyn
1 Shlomi Gutman
1 George Joseph
1 Frank Matano
1 Cédric Bassaget
1 Dan Jenkins
1 Jim Van Meggelen
1 Patrick Wakano
1 Jeremiah Gadd
1 Michael
1 Daniel Heckl
1 Boris P. Korzun
1 Kilburn
1 Bernhard Schmidt
1 Alexander Traud
1 Joeran Vinzens
1 Dennis
1 Vitold
1 Anton Satskiy
1 Kevin Flyn
1 David Cunningham
1 Jim Van Meggelen
1 Vitold
1 Florian Floimair
1 Robert Sutton
1 Daniel
1 Dan Jenkins
1 Ove Aursand
1 Dmitry Wagin
1 Robin Leffmann
1 Mitch Claborn
1 Jonathan Hunter
1 Joshua Roys
1 Olivier Krief
1 Paul Brooks
1 Maciej Michno
1 Kevin Reeves
1 Niklas Larsson
1 Bernhard Schmidt
1 Christoph Moench-Tegeder
1 Maciej Michno
1 Stas Kobzar
1 Cedric BASSAGET
1 EDV O-TON
1 Ted G
1 Frank Matano
1 Yury Kirsanov
1 Anton Satskiy
1 David M. Lee
1 Patrick Wakano
1 Michael Goryainov
1 Niklas Larsson
1 Sebastian Kemper
1 Francois Blackburn
1 Università di Bologna - CESIA VoIP
1 Richard Kenner
1 Niksa Baldun
1 Ian Jones
1 Jean-Denis Girard
1 Dmitriy Serov
1 Peter Turczak
1 Roger James
1 Paulo Vicentini
1 Ted G
1 Martin Zeh
1 Università di Bologna - CESIA VoIP
1 Marin Odrljin
1 Jonas Swiatek
1 Eliel Sardañons
1 AvayaXAsterisk
1 Dirk Wendland
1 Joshua Roys
1 Mark
1 Dan Cropp
1 Jonathan Harris
1 Matt Addison
1 Leandro Dardini
1 alex
1 Chris Savinovich
1 xrobau
1 David Lee
1 Nicholas John Koch
1 Peter Sokolov
1 Eliel Sardañons
1 Sean Bright
1 Aheliotech
1 Bill Kervaski
1 Cyril Ramière
1 Jørgen H
1 Niksa Baldun
1 Kfir Itzhak

Closed Issues

[Back to Top]

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

Security

Category: Channels/chan_sip/General

ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter Addr of a peer
Reported by: Andrey V. T.
    +
  • [4a1cadeadb] Ben Ford -- chan_sip.c: Prevent address change on unauthenticated SIP request.
    • +

Category: Core/ManagerInterface

ASTERISK-28580: Bypass SYSTEM write permission in manager action allows system commands execution
Reported by: Eliel Sardañons
    +
  • [7e3a6e158f] George Joseph -- manager.c: Prevent the Originate action from running the Originate app
    • +

Category: Resources/res_pjsip_t38

ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream causes crash
Reported by: Alexei Gradinari
    +
  • [18f5f5fc99] Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media
    • +

New Feature

Category: Applications/app_senddtmf

ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead of only "sending"
Reported by: lvl
    +
  • [772b59034f] lvl -- app_senddtmf: Add receive mode to AMI Action PlayDTMF
    • +

Category: Core/General

ASTERISK-6863: [patch] allow Asterisk to set high ToS bits as non-root on Linux
Reported by: Matt Addison
    +
  • [a107e85b2e] Alexander Traud -- install_prereq: Add libcap for high bits in DiffServ/ToS.
    • +

Category: Core/Jitterbuffer

ASTERISK-28533: func_jitterbuffer: Add support for video synchronization
Reported by: Joshua C. Colp
    +
  • [7298a785ad] Joshua Colp -- func_jitterbuffer: Add audio/video sync support.
    • +

Category: Functions/func_curl

ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs" doesn't do anything
Reported by: candrews
    +
  • [0c2bf1664c] Sean Bright -- func_curl: Add 'followlocation' option to CURLOPT()
    • +
ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header
Reported by: Martin Tomec
    +
  • [d257a0898e] Martin Tomec -- func_curl.c: Support custom http headers
    • +

Category: Resources/res_musiconhold

ASTERISK-17808: [patch] Unregister a realtime moh class
Reported by: Byron Clark
    +
  • [cf364cd007] sungtae kim -- res_musiconhold: Added unregister realtime moh class
    • +

Category: Resources/res_pjsip_endpoint_identifier_ip

ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on source port
Reported by: Sean Bright
    +
  • [312abaa1fe] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add port matching support
    • +

Category: pjproject/pjsip

ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup From header URI domain
Reported by: Stas Kobzar
    +
  • [c7270dca81] Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN
    • +

Bug

Category: .Release/Targets

ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register
Reported by: Chris Savinovich
    +
  • [172e183b9d] Kevin Harwell -- res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions
    • +

Category: Applications/General

ASTERISK-28954: StreamEcho() only returns 1 active stream
Reported by: Bill Kervaski
    +
  • [00a52b4752] Joshua C. Colp -- app_stream_echo: Fix state of added streams.
    • +
ASTERISK-16676: DAHDIRAS fails to properly initiate pppd unless asterisk is running as root
Reported by: Jaco Kroon
    +
  • [4f92dcd66b] Jaco Kroon -- dahdiras: Only set plugin dahdi.so to pppd if we're running as root.
    • +

Category: Applications/app_amd

ASTERISK-28608: app_amd: Use time calculation to calculate timeout
Reported by: Michael Cargile
    +
  • [5bda460300] Michael Cargile -- app_amd: Fixed timeout issue
    • +

Category: Applications/app_chanisavail

ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
Reported by: Frederic LE FOLL
    +
  • [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.
    • +
ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf
Reported by: Frederic LE FOLL
    +
  • [2d0eee5418] Frederic LE FOLL -- ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf.
    • +

Category: Applications/app_confbridge

ASTERISK-28841: app_confbridge: Add support for disabling text messaging for a user
Reported by: Joshua C. Colp
    +
  • [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling text messaging.
    • +
ASTERISK-28790: Crash during conference call using confbridge and video
Reported by: Pascal Cadotte Michaud
    +
  • [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient space for worst case NACK.
    • +

Category: Applications/app_fax

ASTERISK-28848: app_fax: Compile.
Reported by: Alexander Traud
    +
  • [26b8c99963] Alexander Traud -- app_fax: SpanDSP headers do not use ast_malloc; ignore that.
    • +

Category: Applications/app_meetme

ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
Reported by: George Joseph
    +
  • [ed394ce5b1] Joshua C. Colp -- configure: Add check for MySQL client bool and my_bool type usage.
    • +
  • [a47cb71bb1] George Joseph -- Build: Fix compile issues with seldom used modules
    • +

Category: Applications/app_mixmonitor

ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup
Reported by: Joshua C. Colp
    +
  • [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to attach to hung up channels.
    • +

Category: Applications/app_osplookup

ASTERISK-28804: [patch] app_osplookup.c: Avoid a format truncation.
Reported by: Alexander Traud
    +
  • [527e4f6542] Alexander Traud -- app_osplookup: Avoid a format truncation.
    • +

Category: Applications/app_queue

ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events
Reported by: Ove Aursand
    +
  • [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a call as abandoned
    • +
ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned
Reported by: Kfir Itzhak
    +
  • [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a call as abandoned
    • +
ASTERISK-29034: Lastpause of realtime members is reseting
Reported by: Evandro César Arruda
    +
  • [36dd15c659] Evandro César Arruda -- app_queue: Member lastpause time reseting
    • +
ASTERISK-28951: Inconsistent behaviour queues.conf when there is (not) a [general] section
Reported by: Walter Doekes
    +
  • [312c23b0e1] Walter Doekes -- app_queue: (Breaking change) shared_lastcall and autofill default to no
    • +
ASTERISK-28950: Stale code in app_queue to check untouched channel
Reported by: Walter Doekes
    +
  • [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in try_calling
    • +
ASTERISK-28644: Stale comment in app_queue about ring_entry exception
Reported by: Walter Doekes
    +
  • [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in try_calling
    • +
  • [0e750cdd10] Walter Doekes -- app_queue: Fix old confusing comment about when the members are called
    • +
ASTERISK-28952: Queue wrapuptime sometimes not respected (based on stale lastcall time)
Reported by: Walter Doekes
    +
  • [0fb6738314] Walter Doekes -- app_queue: Read latest wrapuptime instead of (possibly stale) copy
    • +
ASTERISK-28829: app_queue: leaking stasis subscription when Redirecting call
Reported by: lvl
    +
  • [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
    • +
ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
Reported by: Etienne Lessard
    +
  • [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
    • +
ASTERISK-28349: Pause reason not reported in QueueMember AMI event
Reported by: Niksa Baldun
    +
  • [9522390a69] Sean Bright -- app_queue: Deprecate the QueueMemberPause.Reason field
    • +

Category: Applications/app_record

ASTERISK-28682: app_record: Lack of `beep` audio file causes application to return error and hangup
Reported by: Corey Farrell
    +
  • [2f8b20b949] Corey Farrell -- app_record: Do not hang up if beep audio is missing
    • +

Category: Applications/app_system

ASTERISK-28776: Non async-signal-safe syscalls used after fork before exec
Reported by: nappsoft
    +
  • [6b2d945174] Pirmin Walthert -- app.c: make sure that no non-async-signal-safe syscalls are used after
    • +

Category: Applications/app_voicemail

ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command
Reported by: Leandro Dardini
    +
  • [b575868000] Sean Bright -- app_voicemail: Process urgent messages with mailcmd
    • +
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
    +
  • [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail
    • +
ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail to leave message
Reported by: Jim Van Meggelen
    +
  • [9be89d9913] Sean Bright -- app_voicemail: Set globals to default values when voicemail.conf missing
    • +

Category: Applications/app_voicemail/IMAP

ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because not checking mailstream
Reported by: Alexei Gradinari
    +
  • [15624d9a7a] Alexei Gradinari -- app_voicemail/IMAP: check mailstream not NULL in leave_voicemail
    • +

Category: Applications/app_voicemail/ODBC

ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used
Reported by: Stas Kobzar
    +
  • [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving message with realtime voicemail
    • +

Category: Bridges/bridge_builtin_features

ASTERISK-28920: bridge show all causes crash
Reported by: sungtae kim
    +
  • [25ae412f75] sungtae kim -- bridge.c: Fixed null pointer exception
    • +

Category: Bridges/bridge_native_rtp

ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.
Reported by: Frederic LE FOLL
    +
  • [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.
    • +

Category: Bridges/bridge_softmix

ASTERISK-28944: bridge_softmix: Transitioning a stream from inactive -> sendrecv/sendonly doesn't re-negotiation
Reported by: Joshua C. Colp
    +
  • [8ad06394c4] Joshua C. Colp -- bridge_softmix: Add additional old states for adding new source.
    • +
ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp packets
Reported by: Jonathan Hunter
    +
  • [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.
    • +
ASTERISK-28819: [patch] bridge_softmix_binaural: Show state in menuselect.
Reported by: Alexander Traud
    +
  • [7febd22304] Alexander Traud -- bridge_softmix_binaural: Show state in menuselect.
    • +
ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix bridge
Reported by: Kevin Harwell
    +
  • [e77cb32583] Kevin Harwell -- bridge_softmix: clear hold when joining a softmix bridge
    • +

Category: CDR/General

ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
    +
  • [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge
    • +
ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR.
Reported by: Frederic LE FOLL
    +
  • [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail sometimes fails to deactivate CDR.
    • +
ASTERISK-28566: CDR backend unload problem during active call(s)
Reported by: Marian Piater
    +
  • [51850a79ef] Sean Bright -- cdr_mysql: Don't clean up on unload unless we can unregister from CDRs
    • +

Category: CDR/cdr_pgsql

ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column
Reported by: Christoph Moench-Tegeder
    +
  • [52ade18420] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql res_config_pgsql: compatibility with PostgreSQL 12
    • +

Category: Channels/chan_dahdi

ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40
Reported by: Andrew Siplas
    +
  • [5bd7281442] Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to better reflect "no timeout"
    • +
ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
Reported by: Frederic LE FOLL
    +
  • [a68299f508] Frederic LE FOLL -- chan_dahdi: PRI span status may stay "Down, Active" after a short alarm
    • +
ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD
Reported by: Guido Falsi
    +
  • [4072e219f7] Guido Falsi -- chan_dahdi: Fix build with clang/llvm
    • +
ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up
Reported by: Frederic LE FOLL
    +
  • [41b67f150e] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up
    • +

Category: Channels/chan_local

ASTERISK-28938: core_unreal / core_local: Add support for multistream and re-negotiation
Reported by: Joshua C. Colp
    +
  • [de2813cf23] Joshua C. Colp -- core_unreal / core_local: Add multistream and re-negotiation.
    • +
ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
Reported by: Etienne Lessard
    +
  • [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
    • +

Category: Channels/chan_pjsip

ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
Reported by: Joseph Ades
    +
  • [31fbfc5e95] Kevin Harwell -- chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution
    • +
  • [4eba6b9eb2] Kevin Harwell -- PJSIP_MEDIA_OFFER: override configuration on refresh
    • +
ASTERISK-28886: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
Reported by: Jared Smith
    +
  • [8b925fbda3] Kevin Harwell -- chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+
    • +
ASTERISK-28923: T.38 Segfaults in chan_pjsip_queryoption
Reported by: Yury Kirsanov
    +
  • [41f3a7da4d] George Joseph -- res_fax: Don't start a gateway if either channel is hung up
    • +
ASTERISK-28835: IPv6 addresses in SDP incorrectly formatted
Reported by: Daniel Heckl
    +
  • [9f117ac9ef] Daniel Heckl -- res_pjsip: Fixed format of IPv6 addresses for external media addresses
    • +
ASTERISK-28817: chan_pjsip: constant DTMF tone if RTP is not setup yet
Reported by: Kevin Harwell
    +
  • [fa3c8f94e0] Kevin Harwell -- chan_pjsip: digit_begin - constant DTMF tone if RTP is not setup yet
    • +
ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge
Reported by: Michael Neuhauser
    +
  • [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active
    • +
ASTERISK-28759: A non negotiated rtp frame causes call disconnection when there is a SSRC change
Reported by: Paulo Vicentini
    +
  • [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when remote SSRC changes.
    • +
ASTERISK-28766: PJSIP blind transfer not completed after using Proceeding()
Reported by: lvl
    +
  • [d1a2ff0aaf] lvl -- res_pjsip_refer: ensure refer progress is still sent after Proceeding()
    • +
ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field
Reported by: Jean Aunis - Prescom
    +
  • [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI
    • +
ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group
Reported by: Jean-Denis Girard
    +
  • [b40dd11afe] Sean Bright -- res_pjsip_config_wizard: Fix change detection for wizard settings
    • +
ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response Contact
Reported by: Ross Beer
    +
  • [cbc1136704] George Joseph -- res_pjsip_nat: Restore original contact for REGISTER responses
    • +
ASTERISK-28578: race condition on pjsip channelstats command
Reported by: Salah Ahmed
    +
  • [ddb0091da5] Salah Ahmed -- Crash during "pjsip show channelstats" execution
    • +
ASTERISK-28561: Asterisk Deadlocks
Reported by: Aheliotech
    +
  • [bf6f27388d] Joshua Colp -- pbx: deadlock when outgoing dialed channel hangs up too quickly
    • +
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
Reported by: Jeremiah Gadd
    +
  • [c03f50c1c8] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel
    • +
ASTERISK-28538: chan_pjsip: Deadlock on fax detection
Reported by: Joshua C. Colp
    +
  • [c358da472e] Joshua Colp -- chan_pjsip: Relock correct channel during "fax" redirect.
    • +

Category: Channels/chan_sip/General

ASTERISK-29011: chan_sip: ToHost property not cleared on reload
Reported by: Dennis
    +
  • [9058d9e591] Dennis Buteyn -- chan_sip: Clear ToHost property on peer when changing to dynamic host
    • +
ASTERISK-28957: chan_sip: chan_sip does not process 400 response to an INVITE.
Reported by: Frederic LE FOLL
    +
  • [a423f935c9] Frederic LE FOLL -- chan_sip: chan_sip does not process 400 response to an INVITE.
    • +
ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp packets
Reported by: Jonathan Hunter
    +
  • [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.
    • +
ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections
Reported by: Jaco Kroon
    +
  • [365d007eb6] Jaco Kroon -- chan_sip: in case of tcp/tls, be less annoying about tx errors.
    • +
ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP
Reported by: Jean Aunis - Prescom
    +
  • [9c9296c635] Jean Aunis -- chan_sip: voice frames are no longer transmitted after emitting a COLP
    • +
ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility check failure when negociated ptime is not default ptime.
Reported by: Frederic LE FOLL
    +
  • [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no directmedia for ptime other than default ptime.
    • +

Category: Channels/chan_sip/Interoperability

ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should return 503
Reported by: Walter Doekes
    +
  • [43620cbf6c] Walter Doekes -- chan_sip: Return 503 if we're out of RTP ports
    • +
ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed: no audio
Reported by: Walter Doekes
    +
  • [711a3fed56] Walter Doekes -- chan_sip: Always process updated SDP on media source change
    • +

Category: Channels/chan_sip/Messaging

ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan
Reported by: Frank Matano
    +
  • [f309b86e36] Sean Bright -- chan_sip.c: Stop handling continuation lines after reading headers
    • +

Category: Channels/chan_sip/TCP-TLS

ASTERISK-28372: Asterisk REPLY Wrong Contact header port (TCP)
Reported by: Anton Satskiy
    +
  • [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with non-default TCP/TLS ports.
    • +
ASTERISK-24428: Document that Asterisk will use the default SIP ports (5060 for TCP, 5061 for TLS) if the extern option variants aren't used
Reported by: sstream
    +
  • [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with non-default TCP/TLS ports.
    • +
ASTERISK-27195: chan_sip: only sets ToS bits on UDP socket, ignoring TCP and TLS sockets
Reported by: Joshua Roys
    +
  • [4d0ab620be] Alexander Traud -- chan_sip: DiffServ/ToS not only on UDP but also on TCP and TLS sockets.
    • +

Category: Channels/chan_sip/Transfers

ASTERISK-28677: CDR billsec is always 0 for transferred calls
Reported by: Maciej Michno
    +
  • [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when leaving a parking bridge
    • +

Category: Channels/chan_unistim

ASTERISK-28803: [patch] chan_unistim: Avoid tautological warnings with clang.
Reported by: Alexander Traud
    +
  • [b38f664250] Alexander Traud -- chan_unistim: Avoid tautological warnings with clang.
    • +
ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at end of a struct
Reported by: Alexander Traud
    +
  • [3863ab9af9] Igor Goncharovsky -- chan_unistim: Fix clang warning: variable sized type not at end of a struct
    • +

Category: Codecs/codec_resample

ASTERISK-28511: codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32
Reported by: Ruddy G
    +
  • [e4289b9e56] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary
    • +
  • [b096389660] Sean Bright -- codec_resample: Upgrade speex_resample to fix up-sampling bug
    • +

Category: Codecs/codec_silk

ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation' output
Reported by: Sean Bright
    +
  • [dfad69ce7c] Sean Bright -- translate.c: Fix silk 24kHz truncation in 'core show translation'
    • +

Category: Configs/Basic-PBX

ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order mark is present
Reported by: Robin Leffmann
    +
  • [40b5cf8f52] Sean Bright -- config.c: Skip UTF-8 BOMs if present when reading config files
    • +

Category: Configs/Samples

ASTERISK-29123: logger.conf.sample missing comment mark on line 115
Reported by: Andrew Siplas
    +
  • [79d749d2b5] Andrew Siplas -- logger.conf.sample: add missing comment mark
    • +

Category: Contrib/General

ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed to due to invalid syntax
Reported by: Richard Kenner
    +
  • [095c204fe0] snuffy -- contrib/valgrind: Fix use of frame-level suppression
    • +
ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py
Reported by: Pascal Cadotte Michaud
    +
  • [e494d5fd76] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid typo
    • +

Category: Core/ACL

ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault on reading rule from realtime
Reported by: Andrew Yager
    +
  • [7a43bedd72] Sean Bright -- acl.c: Coerce a NULL pointer into the empty string
    • +

Category: Core/Bridging

ASTERISK-28841: app_confbridge: Add support for disabling text messaging for a user
Reported by: Joshua C. Colp
    +
  • [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling text messaging.
    • +

Category: Core/BuildSystem

ASTERISK-28929: pjproject_bundled: Honor --without-pjproject.
Reported by: Alexander Traud
    +
  • [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor --without-pjproject.
    • +
ASTERISK-28837: pjproject_bundled: Honor --without-pjproject.
Reported by: Alexander Traud
    +
  • [966acc6251] Alexander Traud -- pjproject_bundled: Honor --without-pjproject.
    • +
ASTERISK-28824: BuildSystem: Search for Python/C API when possibly needed only.
Reported by: Alexander Traud
    +
  • [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.
    • +
ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.
Reported by: Alexander Traud
    +
  • [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.
    • +
ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
Reported by: Alexander Traud
    +
  • [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and doc/pdf leftovers.
    • +
ASTERISK-28818: [patch] BuildSystem: Allow space in path.
Reported by: Alexander Traud
    +
  • [7a04947abd] Alexander Traud -- BuildSystem: Allow space in path.
    • +
ASTERISK-28487: compile menuselect on gentoo
Reported by: Kilburn
    +
  • [e40f248fac] Sean Bright -- menuselect: Fix curses build on Gentoo Linux
    • +

Category: Core/Channels

ASTERISK-25844: app_queue: Ghost channels in "core show channels" output
Reported by: Etienne Lessard
    +
  • [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in app_queue to avoid leaked stasis subscriptions
    • +
ASTERISK-28795: channel: write to a stream on multi-frame writes
Reported by: Kevin Harwell
    +
  • [3c345ec56d] Kevin Harwell -- channel: write to a stream on multi-frame writes
    • +
ASTERISK-28499: translate: Crash when frame does not have a "src" field set
Reported by: Gregory Massel
    +
  • [1e9714a050] Joshua Colp -- AST-2019-005 - translate: Don't assume all frames will have a src.
    • +

Category: Core/Configuration

ASTERISK-28955: "setvar" doesn't work properly in dahdi-channels.conf
Reported by: Marin Odrljin
    +
  • [d88e230037] Guido Falsi -- chan_dadhi: Fix setvar in dahdi channels
    • +
ASTERISK-23756: setvar directive when used in template and a child of said template, results in duplicate variable names
Reported by: Michael Goryainov
    +
  • [32ce6e9a06] Michael Goryainov -- channels: Allow updating variable value
    • +

Category: Core/General

ASTERISK-28797: [patch] tcptls: Fix notice when TLS is enabled but not configured.
Reported by: Alexander Traud
    +
  • [f9ea75d117] Alexander Traud -- tcptls: Fix notice when TLS is enabled but not supported.
    • +
ASTERISK-28839: Sporadic crashes with Segmentation fault
Reported by: Joeran Vinzens
    +
  • [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP re-negotiation scenarios.
    • +
ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup
Reported by: Joshua C. Colp
    +
  • [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to attach to hung up channels.
    • +
ASTERISK-28498: cel / cdr: Event times may be incorrect
Reported by: Joshua C. Colp
    +
  • [261646c1c4] Joshua Colp -- cdr / cel: Use event time at event creation instead of processing.
    • +

Category: Core/RTP

ASTERISK-28480: json integer overflow in ssrc and timestamp
Reported by: Salah Ahmed
    +
  • [3656c42cb0] Kevin Harwell -- various modules: json integer overflow
    • +

Category: Core/Stasis

ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" field
Reported by: Jean Aunis - Prescom
    +
  • [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text message variables work in ARI
    • +
ASTERISK-28553: stasis.c: Crash during unload
Reported by: Kevin Harwell
    +
  • [729b286d59] Joshua Colp -- stasis: Pass bumped topic_all reference to proxy_dtor.
    • +

Category: Core/Streams

ASTERISK-28870: streams: One memory leak and one issue cloning streams
Reported by: George Joseph
    +
  • [7fbfbe7da0] George Joseph -- streams: Fix one memory leak and one formats ref issue
    • +
ASTERISK-28846: stream: Enforce formats immutability
Reported by: Joshua C. Colp
    +
  • [1c5e68580a] Joshua C. Colp -- stream: Enforce formats immutability and ensure formats exist.
    • +
ASTERISK-28625: Playback of local files impacted by large media cache
Reported by: Kevin Reeves
    +
  • [c626ccec12] Kevin Reeves -- main/file.c: Limit media cache usage to remote files.
    • +

Category: Core/UDPTL

ASTERISK-28483: packet lost on UDPTL wrap around
Reported by: Torrey Searle
    +
  • [084901d548] Torrey Searle -- main/udptl.c: correctly handle udptl sequence wrap around
    • +

Category: Documentation

ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
Reported by: Alexander Traud
    +
  • [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and doc/pdf leftovers.
    • +
ASTERISK-24484: Update documentation for statsd module - usage requirements unclear
Reported by: Dan Jenkins
    +
  • [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does nothing on its own
    • +
ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames
Reported by: Joshua C. Colp
    +
  • [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames
    • +
ASTERISK-28507: Wiki docs missing for MessageWaiting
Reported by: David M. Lee
    +
  • [d5f3ec92d0] George Joseph -- CI: Update buildAsterisk.sh to do a "make full"
    • +

Category: Functions/General

ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation
Reported by: Pascal Cadotte Michaud
    +
  • [bf4dd3d837] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation
    • +
  • [7e3015d779] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing argument documentation
    • +
ASTERISK-26481: FILE function grabs garbage along with read data when target line has no newline
Reported by: Jonathan Harris
    +
  • [bf7c808604] Sean Bright -- func_env: Prevent FILE() from reading garbage at end-of-file
    • +

Category: Functions/func_aes

ASTERISK-28788: func_aes: incorrectly printing error 'declined to load'
Reported by: Alexander Traud
    +
  • [cd8cbf7384] Alexander Traud -- func_aes: Avoid incorrect error message on load.
    • +

Category: Functions/func_channel

ASTERISK-28796: func_channel: cannot read fields exten, context, userfield, channame from dialplan
Reported by: Sébastien Duthil
    +
  • [d40e343710] Sebastien Duthil -- func_channel: allow reading 4 fields from dialplan
    • +

Category: Functions/func_enum

ASTERISK-26711: func_enum: ENUM code wrong case
Reported by: Vitold
    +
  • [517224ce85] Sean Bright -- enum.c: Add support for regular expression flag in NAPTR record
    • +
ASTERISK-19460: [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string
Reported by: George Joseph
    +
  • [ab63f0cd0f] Sean Bright -- enum.c: Make ast_get_txt() actually do something.
    • +

Category: Functions/func_odbc

ASTERISK-20325: Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples.
Reported by: Olivier Krief
    +
  • [c4e0983742] Sean Bright -- func_odbc.conf.sample: Clarify sample documentation
    • +
ASTERISK-28497: func_odbc: truncating Unicode string on readsql
Reported by: Boris P. Korzun
    +
  • [8979921da9] Boris P. Korzun -- func_odbc: acf_odbc_read() and cli_odbc_read() unicode support
    • +

Category: Functions/func_version

ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
Reported by: cmaj
    +
  • [543f936147] cmaj -- Makefile: Fix certified version numbers
    • +

Category: General

ASTERISK-28930: ./configure --without-ssl build failure
Reported by: Jaco Kroon
    +
  • [9b5042433b] Joshua C. Colp -- menuselect: Resolve infinite loop in dependency scenario.
    • +
ASTERISK-28838: AST_MODULE_INFO requires, MODULEINFO does not mention
Reported by: Alexander Traud
    +
  • [abf4d74384] Alexander Traud -- cdr_odbc: Sync load- and build-time deps.
    • +
  • [191f136260] Alexander Traud -- res_pjsip_refer: Add build-time dependency.
    • +
  • [5c2b8fdeca] Alexander Traud -- app_getcpeid: Add build-time dependency.
    • +
  • [008f46bf1e] Alexander Traud -- res_pjsip: Sync load- and build-time deps.
    • +
  • [e2affa3b0a] Alexander Traud -- curl: Add build-time dependency.
    • +
  • [f1135b453b] Alexander Traud -- res_pjsip: Add build-time dependency.
    • +
ASTERISK-28609: Memory Leak in res_rtp_asterisk.c
Reported by: Ted G
    +
  • [39c920ac78] George Joseph -- res_rtp_asterisk: Add frame list cleanups to ast_rtp_read
    • +
ASTERISK-28590: utils.c throws repeated warnings; "pthread_attr_setstacksize: Invalid argument"
Reported by: Speed Dial Dave
    +
  • [a4222614c4] Sean Bright -- utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN
    • +
ASTERISK-28523: Asterisk 16.5.0 Memory leak
Reported by: Cyril Ramière
    +
  • [a4caaef64c] Kevin Harwell -- res_sorcery_memory_cache: stale item update leak
    • +
ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to srtp_protect/unprotect causing SEGV
Reported by: Jonas Swiatek
    +
  • [b805e1237d] Kevin Harwell -- srtp: Fix possible race condition, and add NULL checks
    • +

Category: PBX/General

ASTERISK-29046: pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension
Reported by: Ramarajan
    +
  • [6d50d152d8] Joshua C. Colp -- pbx: Fix hints deadlock between reload and ExtensionState.
    • +
ASTERISK-28695: core: minmemfree watermark uses free RAM, not available RAM
Reported by: Kevin Flyn
    +
  • [50d02d6194] Sean Bright -- pbx.c: Include filesystem cache in free memory calculation
    • +
ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X
Reported by: Dirk Wendland
    +
  • [ee7d72eb72] George Joseph -- sig_pri: Fix deadlock caused by sig_pri_queue_hangup
    • +

Category: PBX/pbx_config

ASTERISK-28534: Segmentation fault when there is no priority for an extension
Reported by: Timothy Vanderaerden
    +
  • [702019fc80] Sean Bright -- pbx: Prevent Realtime switch crash on invalid priority
    • +

Category: PBX/pbx_dundi

ASTERISK-21205: [patch] dundi_read_result crash due to negative number
Reported by: Jaco Kroon

Category: Resources/res_ari

ASTERISK-28948: ARI channel create doesn't referencing the channel_id parameter
Reported by: sungtae kim
    +
  • [bbe0f2230d] sungtae kim -- res_ari: Fix create channel request channelId parameter parsing
    • +
ASTERISK-28679: stasis application is destroyed after its creation
Reported by: Francois Blackburn
    +
  • [4206830a52] Kevin Harwell -- res_stasis: trigger cleanup after update
    • +
ASTERISK-28585: ari/resource_events: Crash in event session cleanup
Reported by: Kevin Harwell
    +
  • [360936ead5] Joshua Colp -- res_ari_events: Add module reference when a WebSocket is open.
    • +

Category: Resources/res_ari_bridges

ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp packets
Reported by: Jonathan Hunter
    +
  • [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio from mixed frame.
    • +

Category: Resources/res_ari_channels

ASTERISK-28940: /channels/create doesn't get any parameters from the body
Reported by: sungtae kim
    +
  • [fa7c69f40f] sungtae kim -- res_ari: Fix create request body parameter parsing.
    • +
ASTERISK-28847: ARI channels cuts the endpoint string over 80 characters
Reported by: sungtae kim
    +
  • [9ad3d2829c] sungtae kim -- res_ari_channels: Fixed endpoint 80 characters limit
    • +

Category: Resources/res_calendar_exchange

ASTERISK-28572: Memory leaks in res_calendar_exchange and res_calendar_icalendar
Reported by: Yoooooo Ha
    +
  • [16e668c7dd] Sean Bright -- res_calendar: Resolve memory leak on calendar destruction
    • +

Category: Resources/res_calendar_icalendar

ASTERISK-28572: Memory leaks in res_calendar_exchange and res_calendar_icalendar
Reported by: Yoooooo Ha
    +
  • [16e668c7dd] Sean Bright -- res_calendar: Resolve memory leak on calendar destruction
    • +

Category: Resources/res_corosync

ASTERISK-28888: res_corosync: causes asterisk crash in huge distributed environment.
Reported by: Università di Bologna - CESIA VoIP
    +
  • [0c1c386634] Università di Bologna - CESIA VoIP -- res_corosync: Fix crash in huge distributed environment.
    • +

Category: Resources/res_fax

ASTERISK-28900: res_fax: Double frame free when gateway in use with off-nominal format usage
Reported by: Gregory Massel
    +
  • [d2500c6273] Joshua C. Colp -- res_fax: Don't consume frames given to fax gateway on write.
    • +
ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config option
Reported by: Kevin Harwell
    +
  • [b6f5607359] Kevin Harwell -- res_fax: wrap v21 detected Asterisk initiated negotiation with config option
    • +

Category: Resources/res_http_websocket

ASTERISK-28975: res_http_websocket: Text payload data doesn't necessary include trailing zero
Reported by: Nickolay V. Shmyrev
    +
  • [e4d24f5137] Nickolay Shmyrev -- res_http_websocket: Avoid reading past end of string
    • +
ASTERISK-28562: SIP WSS message not processed until next frame arrives
Reported by: Robert Sutton
    +
  • [87110c1bdf] Sean Bright -- websocket: Consider pending SSL data when waiting for socket input
    • +

Category: Resources/res_musiconhold

ASTERISK-28927: Asterisk crash in music on hold
Reported by: David Cunningham
    +
  • [57554c2834] Sean Bright -- res_musiconhold.c: Prevent crash with realtime MoH
    • +
ASTERISK-28892: res_musiconhold: Module res_musiconhold throws false warning
Reported by: Nicholas John Koch
    +
  • [fef97a9a72] Nicholas John Koch -- res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings
    • +
ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN
Reported by: Ross Beer
    +
  • [aeff1f2c53] Sean Bright -- res_musiconhold: Avoid spurious warning when 'format' is the empty string
    • +

Category: Resources/res_parking

ASTERISK-29042: res_parking: Parker UUID is no longer copied
Reported by: Misha Vodsedalek
    +
  • [4f0766dcda] Joshua C. Colp -- parking: Copy parker UUID as well.
    • +
ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the same
Reported by: Ross Beer
    +
  • [811ae88da4] Joshua Colp -- parking: Fall back to parker channel name even if it matches parkee.
    • +
ASTERISK-28616: parking: Deadlock when multi call parking
Reported by: Joshua C. Colp
    +
  • [807a70b7ae] Joshua Colp -- parking: Fix case where we can't get the parker.
    • +
  • [e924c5107c] Joshua Colp -- parking: Use channel snapshot instead of channel.
    • +

Category: Resources/res_pjsip

ASTERISK-28995: res_pjsip_registrar: Expires on statically configured contacts is not correct
Reported by: tootai
    +
  • [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an expiration for static contacts.
    • +
ASTERISK-28965: res_pjsip: Apply outbound proxy to static contacts on AOR
Reported by: Joshua C. Colp
    +
  • [4f86118bd8] Joshua C. Colp -- res_pjsip: Apply AOR outbound proxy to static contacts.
    • +
ASTERISK-28936: res_pjsip: crash when dialing non-sip uri
Reported by: Walter Doekes
    +
  • [e74dde5100] Walter Doekes -- pjsip: Prevent invalid memory access when attempting to contact a non-sip URI
    • +
ASTERISK-28794: res_pjsip: Crash when escaping during URI printing
Reported by: nappsoft
    +
  • [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing the contact_user value.
    • +
ASTERISK-26780: res_pjsip: PJSIP Registration Fails when transport=transport-udp6
Reported by: Peter Sokolov
    +
  • [c8dec423d2] Peter Sokolov -- pjsip_resolver.c: Ensure AAAA dns requests are made.
    • +
ASTERISK-28854: SIGSEGV when pjsip show history encounters IPV6 address
Reported by: Roger James
    +
  • [4a072c4890] Roger James -- res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered.
    • +
ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint synchronization for a specific AOR
Reported by: Jason Hord
    +
  • [d845464c76] Jason Hord -- res_pjsip: Don't set endpoint to unavailable in all cases.
    • +
ASTERISK-28790: Crash during conference call using confbridge and video
Reported by: Pascal Cadotte Michaud
    +
  • [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient space for worst case NACK.
    • +
ASTERISK-28743: Asterisk is crashing if the 200 OK with SDP
Reported by: sungtae kim
    +
  • [8147f43756] Sungtae Kim -- res_pjsip_session: Fixed wrong session termination
    • +
ASTERISK-23407: Fix the FSF address in the headers of lots of pjproject files
Reported by: Jared Smith
    +
  • [0a7fe3097f] Jared Smith -- indications.conf.sample: Add indication tones for Indonesia
    • +
ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls
Reported by: Paul Brooks
    +
  • [313189aae2] Sean Bright -- chan_pjsip: Ignore RTP that we haven't negotiated
    • +
ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR points to a not existent AOR
Reported by: Ross Beer
    +
  • [b1be06df8d] Sean Bright -- res_pjsip_registrar.c: Prevent potential double free if AOR is not found
    • +
ASTERISK-28544: Wrong contact representation in ipv6 mode
Reported by: Jørgen H
    +
  • [377d7bdab6] Sean Bright -- res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6
    • +
ASTERISK-28521: pjsip: Memory Leak
Reported by: Mark
    +
  • [cc83e76aa5] George Joseph -- pjproject_bundled: Revert pjproject 2.9 commits causing leaks
    • +
ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries
Reported by: Ian Jones
    +
  • [86452c9fa4] Joshua Colp -- res_pjsip: Fix multiple of the same contact in "pjsip show contacts".
    • +

Category: Resources/res_pjsip_acl

ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed
Reported by: Timothy Vanderaerden
    +
  • [d6712790cd] Joshua C. Colp -- pjsip: Update ACLs on named ACL changes.
    • +

Category: Resources/res_pjsip_diversion

ASTERISK-29001: chan_pjsip does not process or forward 181 responses
Reported by: Torrey Searle
    +
  • [addd295cda] Torrey Searle -- res_pjsip_diversion: handle 181
    • +

Category: Resources/res_pjsip_endpoint_identifier_ip

ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for hostnames
Reported by: Joshua C. Colp
    +
  • [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document support for hostnames
    • +

Category: Resources/res_pjsip_logger

ASTERISK-28932: res_pjsip_logger writing too big packets
Reported by: nappsoft
    +
  • [e8c6e9ae5d] Pirmin Walthert -- res_pjsip_logger: use the correct pointer when logging tx_messages to pcap
    • +
ASTERISK-28921: Wrong return value check for fwrite when writing to pcap file
Reported by: nappsoft
    +
  • [c16937cdbe] Pirmin Walthert -- res_pjsip_logger.c: correct the return value checks when writing to pcap
    • +

Category: Resources/res_pjsip_messaging

ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be changed
Reported by: Alex
    +
  • [03d24ca4c1] Sean Bright -- res_pjsip_messaging: Allow Content-Type to be overridden
    • +
ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending
Reported by: Dmitriy Serov
    +
  • [b1ca2c5d71] Sean Bright -- res_pjsip_messaging: Ensure MESSAGE_SEND_STATUS is set properly
    • +

Category: Resources/res_pjsip_mwi

ASTERISK-28575: MWI Send Notify Crash on 16.6
Reported by: Joshua Elson
    +
  • [5dae803eea] Kevin Harwell -- res_pjsip_mwi: potential double unref, and potential unwanted double link
    • +
ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi container
Reported by: Kevin Harwell
    +
  • [12dbeb69b0] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object for mwi containers
    • +

Category: Resources/res_pjsip_nat

ASTERISK-28884: x-ast-orig-host not filtered out from request URI and To header
Reported by: nappsoft
    +
  • [1399f8b4fe] Pirmin Walthert -- res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header
    • +

Category: Resources/res_pjsip_notify

ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present instead of just one
Reported by: AvayaXAsterisk
    +
  • [90af050fa4] Sean Bright -- res_pjsip_notify: Only allow a single Event header to be added to a NOTIFY
    • +

Category: Resources/res_pjsip_outbound_registration

ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set
Reported by: George Joseph
    +
  • [78b01f41ae] George Joseph -- res_pjsip_outbound_registration: Fix SRV failover on timeout
    • +
ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover
Reported by: Kevin Harwell
    +
  • [d5d41409e2] Kevin Harwell -- res_pjsip_outbound_registration: add support for SRV failover
    • +
ASTERISK-28521: pjsip: Memory Leak
Reported by: Mark
    +
  • [cc83e76aa5] George Joseph -- pjproject_bundled: Revert pjproject 2.9 commits causing leaks
    • +

Category: Resources/res_pjsip_path

ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured
Reported by: Juan Martin
    +
  • [982a5025b3] Sean Bright -- res_pjsip_registrar: Validate Contact URI before adding to responses
    • +

Category: Resources/res_pjsip_pubsub

ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults
Reported by: Ross Beer
    +
  • [a1f0c833ab] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence data ref when recreating.
    • +
ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not preserve XML version number
Reported by: Bryan Nelson
    +
  • [4e7adbd8f4] Joshua C. Colp -- res_pjsip_pubsub: Add ability to persist generator state information.
    • +

Category: Resources/res_pjsip_registrar

ASTERISK-28995: res_pjsip_registrar: Expires on statically configured contacts is not correct
Reported by: tootai
    +
  • [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an expiration for static contacts.
    • +

Category: Resources/res_pjsip_sdp_rtp

ASTERISK-28784: res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
Reported by: Joshua C. Colp
    +
  • [34750d2068] Joshua C. Colp -- res_pjsip_sdp_rtp: Only do hold/unhold on default audio stream.
    • +
ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during direct-media (native_rtp) bridge
Reported by: Michael Neuhauser
    +
  • [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore rtptimeout if direct-media is active
    • +
ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold
Reported by: Ross Beer
    +
  • [77c9ba8e63] Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH transitions
    • +
ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used
Reported by: Torrey Searle
    +
  • [bf4340f0ec] Torrey Searle -- res_pjsip_sdp_rtp: implement hold state handling on moh_passthrough
    • +
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
    +
  • [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.
    • +

Category: Resources/res_pjsip_session

ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
Reported by: Ross Beer
    +
  • [5cc4a391b3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call pref.
    • +
ASTERISK-29033: res_pjsip_session: Aggressively terminates session on failed re-INVITE
Reported by: Joshua C. Colp
    +
  • [3c074038fe] Joshua C. Colp -- res_pjsip_session: Don't aggressively terminate on failed re-INVITE.
    • +
ASTERISK-28953: res_pjsip_session: Preserve stream label
Reported by: Joshua C. Colp
    +
  • [ee8ea9275f] Joshua C. Colp -- res_pjsip_session: Preserve label on incoming re-INVITE.
    • +
ASTERISK-28871: res_pjsip_session: Unnecessary re-Invite on call answer
Reported by: Alexei Gradinari
    +
  • [afa2c9a868] Joshua C. Colp -- bridge: Don't try to match audio formats.
    • +
ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have reflected state
Reported by: Joshua C. Colp
    +
  • [9620ecbf80] Joshua C. Colp -- res_pjsip_session: Don't restrict non-audio default streams to sendrecv.
    • +
ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes
Reported by: Joshua C. Colp
    +
  • [ac155decae] Joshua C. Colp -- res_pjsip_session: Fix off-nominal session refreshes.
    • +
ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them
Reported by: nappsoft
    +
  • [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on created streams for incoming SDP.
    • +
ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on hangup when TEST_FRAMEWORK enabled
Reported by: Bernhard Schmidt
    +
  • [6ee1f1f507] Sean Bright -- res_pjsip_session.c: Prevent use-after-free with TEST_FRAMEWORK enabled
    • +
ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI
Reported by: Jeremiah Gadd
    +
  • [c03f50c1c8] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF on hungup channel
    • +

Category: Resources/res_pjsip_t38

ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received
Reported by: Salah Ahmed
    +
  • [330ffa2bce] Salah Ahmed -- res_pjsip_t38: T.38 error correction mode selection at 200 ok received
    • +

Category: Resources/res_realtime

ASTERISK-21794: CLI command 'realtime update2' syntax failure when using according to usage help
Reported by: Cedric BASSAGET
    +
  • [094e87b0dc] Sean Bright -- res_realtime: Fix 'realtime update2' argument handling
    • +

Category: Resources/res_rtp_asterisk

ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string to each message block.
Reported by: Thomas Johnson
    +
  • [5ec7099312] Sean Bright -- bridge_channel: Ensure text messages are zero terminated
    • +
ASTERISK-28939: res_rtp_asterisk: Don't have send/receive buffers on non-WebRTC
Reported by: Joshua C. Colp
    +
  • [c84d962eae] Joshua C. Colp -- res_rtp_asterisk: Don't assume setting retrans props means to enable.
    • +
ASTERISK-28904: RTP ICE leaks the memory
Reported by: sungtae kim
    +
  • [c8c94b6cf1] sungtae kim -- res_rtp_asterisk.c: Fixed memory leak
    • +
ASTERISK-28852: Unprotected access to nochecksums variable, causes build failures
Reported by: Guido Falsi
    +
  • [e4366308e1] Guido Falsi -- res_rtp_asterisk: Protect access to nochecksums with #ifdef
    • +
ASTERISK-28827: res_rtp_asterisk: Loop when receive buffer is flushed by a received packet that is also in receive buffer with NACK
Reported by: nappsoft
    +
  • [d50fd0acc0] Pirmin Walthert -- res_rtp_asterisk: Resolve loop when receive buffer is flushed
    • +
ASTERISK-28826: res_rtp_asterisk: Duplicate seqnos being added to send buffer with NACK
Reported by: nappsoft
    +
  • [ca032d1e2e] Pirmin Walthert -- res_rtp_asterisk: Free payload when error on insertion to data buffer
    • +
ASTERISK-28812: First DTMF is not get
Reported by: Bernard Merindol
    +
  • [7db03e12a7] Bernard Merindol -- res_rtp_asterisk.c: Check for first DTMF having timestamp set to 0
    • +
ASTERISK-28809: [patch] res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
Reported by: Alexander Traud
    +
  • [1ef1b1b0c2] Alexander Traud -- res_rtp_asterisk: Avoid absolute value on unsigned subtraction.
    • +
ASTERISK-28773: Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
Reported by: Torrey Searle
    +
  • [a1dba820cf] Torrey Searle -- res_rtp_asterisk: Send correct sender SSRC when p2p bridge in use
    • +
ASTERISK-28769: DTLS Handshake Fails to Occur if ice_support is enabled but not used
Reported by: Torrey Searle
    +
  • [14ba1806f3] Torrey Searle -- res_pjsip_sdp_rtp: Don't wait for ICE if not negotiated
    • +
ASTERISK-28759: A non negotiated rtp frame causes call disconnection when there is a SSRC change
Reported by: Paulo Vicentini
    +
  • [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when remote SSRC changes.
    • +
ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling
Reported by: Joshua C. Colp
    +
  • [87fda066ea] Joshua C. Colp -- res_rtp_asterisk: Improve video performance in certain networks.
    • +
ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before allowing sending
Reported by: Benjamin Keith Ford
    +
  • [168637cc0c] Ben Ford -- RTP/ICE: Send on first valid pair.
    • +
ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup
Reported by: Kevin Harwell
    +
  • [3865b3fd6a] Kevin Harwell -- res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup
    • +
ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet length doesn't match
Reported by: Joshua Elson
    +
  • [02129ad4d0] Joshua Colp -- res_rtp_asterisk: Always return provided DTLS packet length.
    • +

Category: Resources/res_sorcery_memory_cache

ASTERISK-28942: res_sorcery_memory_cache: Individual object expiration behaves unexpectedly with full backend caching
Reported by: Joshua C. Colp
    +
  • [a143c3a7b7] Joshua C. Colp -- res_sorcery_memory_cache: Disallow per-object expire with full backend.
    • +

Category: Resources/res_speech

ASTERISK-29040: res_speech: Assertion on format
Reported by: Nickolay V. Shmyrev
    +
  • [0319e0b07f] Nickolay Shmyrev -- res_speech: Bump reference on format object
    • +

Category: Resources/res_srtp

ASTERISK-28903: res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
Reported by: Alexander Traud
    +
  • [4de0e50c32] Alexander Traud -- res_srtp: Set all possible flags while selecting the Crypto Suite.
    • +
ASTERISK-22920: Crash while Forwarding from TLS extension with CHANNEL args secure_bridge_media and secure_bridge_signaling
Reported by: Shlomi Gutman
    +
  • [29070b61f7] Alexander Traud -- core_local: Local calls are always secure.
    • +

Category: Resources/res_stasis

ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info
Reported by: sungtae kim
    +
  • [2e32b56bdb] sungtae kim -- stasis_bridge.c: Fixed wrong video_mode shown
    • +
ASTERISK-28423: ARI causes STASIS Deadlock
Reported by: Ross Beer
    +
  • [cce2b0da95] Kevin Harwell -- stasis/app: don't lock an app before a call to send
    • +
  • [990a91b44a] George Joseph -- stasis: Don't hold app_registry and session locks unnecessarily
    • +
ASTERISK-28633: stasis bridge topic leak
Reported by: Joeran Vinzens
    +
  • [1c9ddad4db] George Joseph -- stasis.c: Use correct topic name in stasis_topic_pool_delete_topic
    • +

Category: Resources/res_stasis_playback

ASTERISK-28713: res_stasis_playback: Error building JSON
Reported by: Sébastien Duthil
    +
  • [31dc904380] Sean Bright -- res_stasis_playback: Prevent media_index from going out of bounds
    • +

Category: Resources/res_statsd

ASTERISK-24484: Update documentation for statsd module - usage requirements unclear
Reported by: Dan Jenkins
    +
  • [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does nothing on its own
    • +

Category: Tests/General

ASTERISK-28808: [patch] test_stasis: Avoid always true warning with clang.
Reported by: Alexander Traud
    +
  • [bb28ed0d1b] Alexander Traud -- test_stasis: Avoid always true warning with clang.
    • +

Category: Tests/testsuite

ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming Language is python-2.7.
Reported by: Alexander Traud
    +
  • [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API when possibly needed only.
    • +
ASTERISK-28789: test_utils: incorrectly printing error 'declined to load'
Reported by: Alexander Traud
    +
  • [fc07eeaba1] Alexander Traud -- test_utils: Avoid incorrect error message on load.
    • +

Category: Utilities/General

ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling troubles
Reported by: Sebastian Kemper
    +
  • [b7fbb9c41f] Sebastian Kemper -- check_expr2: fix cross-compile/hardening issues
    • +

Category: pjproject/pjsip

ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address)
Reported by: Michael Neuhauser
    +
  • [6482ab5bea] Michael Neuhauser -- pjproject: clone sdp to protect against (nat) modifications
    • +
ASTERISK-28929: pjproject_bundled: Honor --without-pjproject.
Reported by: Alexander Traud
    +
  • [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor --without-pjproject.
    • +
ASTERISK-28794: res_pjsip: Crash when escaping during URI printing
Reported by: nappsoft
    +
  • [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing the contact_user value.
    • +
ASTERISK-28859: pjsip: Increase maximum candidate count
Reported by: Joshua C. Colp
    +
  • [3078a00a6d] Joshua C. Colp -- pjsip: Increase maximum ICE candidate count.
    • +
ASTERISK-28811: Crash occurs when fax session switches from T.38 to audio
Reported by: Alexey Vasilyev
    +
  • [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP re-negotiation scenarios.
    • +
ASTERISK-28837: pjproject_bundled: Honor --without-pjproject.
Reported by: Alexander Traud
    +
  • [966acc6251] Alexander Traud -- pjproject_bundled: Honor --without-pjproject.
    • +
ASTERISK-28758: pjsip startup errors when using "with-ssl" configure option
Reported by: Patrick Wakano
    +
  • [3431949a52] Alexander Traud -- pjproject_bundled: Repair ./configure --with-ssl without ARG.
    • +
ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6 Address Delimited by "[]" Rejected
Reported by: Peter Sokolov
    +
  • [9d9bde76a9] Sean Bright -- pjproject_bundled: Allow brackets in via parameters
    • +
ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5
Reported by: Niklas Larsson
    +
  • [5d9f9f4871] George Joseph -- pjproject_bundled: Replace earlier reverts with official fixes.
    • +
ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters, NEC only supports up to 32 characters
Reported by: Dan Cropp
    +
  • [0844d6b127] Dan Cropp -- pjproject: Configurable setting for cnonce to include hyphens or not
    • +

Improvement

Category: Addons/chan_mobile

ASTERISK-28832: chan_mobile creates PCMA streams that make some VoIP clients crash or not render received audio
Reported by: Peter Turczak
    +
  • [3303defd3f] Peter Turczak -- chan_mobile: Add smoother to make SIP/RTP endpoints happy.
    • +

Category: Applications/NewFeature

ASTERISK-28484: Add AudioSocket support
Reported by: Seán C. McCord
    +
  • [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, and ARI support.
    • +

Category: Applications/app_confbridge

ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. Colp
    +
  • [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.
    • +

Category: Applications/app_mixmonitor

ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
Reported by: xrobau
    +
  • [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
    • +

Category: Applications/app_page

ASTERISK-27946: dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't
Reported by: Joshua Elson
    +
  • [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character limitation
    • +

Category: Applications/app_voicemail

ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.
Reported by: Michael
    +
  • [7362647e2f] Sean Bright -- Revert "app_voicemail: Cleanup stale lock files on module load"
    • +

Category: Applications/app_voicemail/ODBC

ASTERISK-22192: [patch] Allow voicemail forwards with ODBC backend when format differs from attachfmt column
Reported by: cmaj
    +
  • [2d67dbfef5] cmaj -- app_voicemail.c: Support multiple file formats for forwarded messages.
    • +

Category: Bridges/bridge_native_rtp

ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
    +
  • [5a5be92b79] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
    • +

Category: Bridges/bridge_simple

ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
    +
  • [5a5be92b79] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
    • +

Category: Bridges/bridge_softmix

ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
    +
  • [5a5be92b79] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
    • +
ASTERISK-28658: app_confbridge: Add support for setting maximum sample rate
Reported by: Joshua C. Colp
    +
  • [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying maximum sample rate.
    • +

Category: Channels/NewFeature

ASTERISK-28484: Add AudioSocket support
Reported by: Seán C. McCord
    +
  • [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, and ARI support.
    • +

Category: Channels/chan_pjsip

ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail
Reported by: cmaj
    +
  • [fe3cce816c] Richard Mudgett -- app_chanisavail.c: Simplify dialplan using ChanIsAvail.
    • +
  • [abcb4ab321] Richard Mudgett -- app_dial.c: Simplify dialplan using Dial.
    • +
  • [d86a6ac5ce] Richard Mudgett -- app_page.c: Simplify dialplan using Page.
    • +

Category: Contrib/General

ASTERISK-28726: install_prereq script uses the interactive mode when installing aptitude
Reported by: Sylvain Afchain
    +
  • [0c02d0a450] Sylvain Afchain -- install_prereq: Install aptitude non-interactively
    • +

Category: Core/CodecInterface

ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec
Reported by: Florian Floimair
    +
  • [c18983207d] Florian Floimair -- core: Add H.265/HEVC passthrough support
    • +

Category: Core/DNS

ASTERISK-28853: Missing include on FreeBSD
Reported by: Guido Falsi
    +
  • [97494d8984] Guido Falsi -- core/dns: Add system include required on FreeBSD
    • +

Category: Core/Dial

ASTERISK-27946: dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't
Reported by: Joshua Elson
    +
  • [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character limitation
    • +

Category: Core/HTTP

ASTERISK-28750: TLS/SSL Key too small error
Reported by: Martin Zeh
    +
  • [7f2d56fc8c] Sean Bright -- tcptls.c: Log more informative OpenSSL errors
    • +
ASTERISK-28710: Should be able to disable the /httpstatus URI in the built-in HTTP server
Reported by: Sean Bright
    +
  • [0dce6f746b] Sean Bright -- http: Add ability to disable /httpstatus URI
    • +

Category: Core/ManagerInterface

ASTERISK-28945: AMI SendText - add Content-Type parameter
Reported by: Kevin Harwell
    +
  • [cfed0ea033] Kevin Harwell -- manager - Add Content-Type parameter to the SendText action
    • +

Category: Core/Streams

ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
    +
  • [5a5be92b79] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
    • +

Category: Documentation

ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor
Reported by: xrobau
    +
  • [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to correct value when wav49 is used
    • +
ASTERISK-28673: GET FULL VARIABLE documentation clarification
Reported by: Jonathan Harris
    +
  • [7d94bdde9d] Sean Bright -- res_agi: Improve GET FULL VARIABLE documentation
    • +
ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md
Reported by: Sam Banks
    +
  • [0dc7e29dd8] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling correetions.
    • +

Category: Functions/func_volume

ASTERISK-28813: func_volume: Allow decimal numbers as parameter to improve granularity
Reported by: Jean Aunis - Prescom
    +
  • [de66713fd5] Jean Aunis -- func_volume: Accept decimal number as argument
    • +

Category: Resources/NewFeature

ASTERISK-28484: Add AudioSocket support
Reported by: Seán C. McCord
    +
  • [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, and ARI support.
    • +

Category: Resources/res_ari_bridges

ASTERISK-28629: [patch] Add an "inhibitCOLP" flag to the bridges REST API
Reported by: Jean Aunis - Prescom
    +
  • [034ac357ad] Jean Aunis -- ARI: Ability to inhibit COLP frames when adding channels to a bridge
    • +

Category: Resources/res_ari_channels

ASTERISK-28896: ari: Add support for specifying variables on channel create
Reported by: Joshua C. Colp
    +
  • [15cbff9d54] Joshua C. Colp -- ari: Allow variables to be set on channel create.
    • +

Category: Resources/res_http_websocket

ASTERISK-28958: Continue reading string when ping received by websocket
Reported by: Nickolay V. Shmyrev
    +
  • [7163efd934] Nickolay Shmyrev -- res_http_websocket.c: Continue reading after ping/pong
    • +
ASTERISK-28949: res_http_websocket: Add masking to websocket client
Reported by: Moises Silva
    +
  • [9445dac43b] Moises Silva -- res_http_websocket: Add payload masking to the websocket client
    • +

Category: Resources/res_pjsip

ASTERISK-28959: res_pjsip: Added option for disable rport parameter set
Reported by: sungtae kim
    +
  • [81b5e4a73f] sungtae kim -- res_pjsip.c: Added disable_rport option for pjsip.conf
    • +
ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
Reported by: Kevin Harwell
    +
  • [2ee455958e] George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref
    • +
ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
Reported by: Kevin Harwell
    +
  • [06dada3f01] Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option
    • +
ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold re-invites
Reported by: Torrey Searle
    +
  • [b43cdc7f1e] Torrey Searle -- channel/chan_pjsip: add dialplan function for music on hold
    • +

Category: Resources/res_pjsip_logger

ASTERISK-28895: res_pjsip_logger: Add tons'o'functionality
Reported by: Joshua C. Colp
    +
  • [a7aaee70c6] Joshua C. Colp -- res_pjsip_logger: Expand functionality to improve logging.
    • +

Category: Resources/res_pjsip_outbound_registration

ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached
Reported by: Daniel
    +
  • [e73eba85c1] Joshua Colp -- res_pjsip_outbound_registration: Extend documentation for "max_retries".
    • +

Category: Resources/res_pjsip_sdp_rtp

ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
Reported by: Kevin Harwell
    +
  • [2ee455958e] George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref
    • +
ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
Reported by: Kevin Harwell
    +
  • [06dada3f01] Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option
    • +
ASTERISK-28733: stream: Add support for adding/removing streams during SFU/calls
Reported by: Joshua C. Colp
    +
  • [5a5be92b79] Joshua C. Colp -- bridging: Add better support for adding/removing streams.
    • +

Category: Resources/res_pjsip_session

ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option
Reported by: Kevin Harwell
    +
  • [2ee455958e] George Joseph -- codec_negotiation: Implement outgoing_call_offer_pref
    • +
ASTERISK-28782: Add support for Content-Disposition header in multi-part INVITES
Reported by: Torrey Searle
    +
  • [e12244153a] Torrey Searle -- res_pjsip_session: implement processing of Content-Disposition
    • +
ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add video
Reported by: Joshua C. Colp
    +
  • [21e9051461] Joshua C. Colp -- res_pjsip_session: Apply intention behind requested formats.
    • +
ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option
Reported by: Kevin Harwell
    +
  • [06dada3f01] Kevin Harwell -- codec negotiation: add incoming_call_offer_prefs option
    • +

Category: Third-Party/pjproject

ASTERISK-28866: third-party/pjproject/configure.m4 contains bashisms
Reported by: Guido Falsi
    +
  • [c831f03273] Guido Falsi -- pjproject: Remove bashism from configure.m4 script
    • +

Category: pjproject/pjsip

ASTERISK-28899: Upgrade Asterisk to bundled pjproject 2.10
Reported by: Kevin Harwell
    +
  • [415b55af5a] Kevin Harwell -- pjproject: Upgrade bundled version to pjproject 2.10
    • +
ASTERISK-28879: pjproject has race conditions in it's build system
Reported by: Guido Falsi
    +
  • [801d570f6e] Guido Falsi -- pjproject: Fix race condition when building with parallel make
    • +

Open Issues

[Back to Top]

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

Bug

Category: Channels/chan_sip/TCP-TLS

ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
Reported by: Alexander Traud
    +
  • [da9554d925] Alexander Traud -- chan_sip: TCP/TLS client without server.
    • +

Category: Core/Configuration

ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues
Reported by: EDV O-TON
    +
  • [eb9252ea27] Sean Bright -- res_config_odbc: Preserve empty strings returned by the database
    • +

Commits Not Associated with an Issue

[Back to Top]

This is a list of all changes that went into this release that did not reference a JIRA issue.

+ + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + +
RevisionAuthorSummary
6fd94258f8Asterisk Development TeamUpdate for 18.0.0-rc2
704cb88799Asterisk Development TeamUpdate for 18.0.0-rc1
f589985840Asterisk Development TeamUpdate CHANGES and UPGRADE.txt for 18.0.0
5a49757e40Patrick Verzeleres_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly
ec03909831Kevin Harwellconversions: Add string to signed integer conversion functions
e32815dddbGeorge Josephast_coredumper: Fix issues with naming
9ed1b1452dAlexander Traudsip_nat_settings: Update script for latest Linux.
217449a1e5Alexander Traudsamples: Fix keep_alive_interval default in pjsip.conf.
5a8cacb93dGeorge Josephlogger.c: Added a new log formatter called "plain"
5dfeeba623Sean Brightres_musiconhold.c: Use ast_file_read_dir to scan MoH directory
c4c72d55a2George Josephscope_trace: Added debug messages and added additional macros
d26ab7f8f9George Josephstream.c: Added 2 more debugging utils and added pos to stream string
6faf76308dGeorge JosephACN: Changes specific to the core
a3d87f78edJoshua C. Colpres_pjsip: Fix codec preference defaults.
da8a617dc9Sean Brightvector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
769a9611e7Ben Fordutils.c: NULL terminate ast_base64decode_string.
802aa97fa0George JosephACN: Configuration renaming for pjsip endpoint
de23cb4002Ben Fordres_stir_shaken: Fix memory allocation error in curl.c
71446b68fcGeorge Josephres_pjsip_session: Ensure reused streams have correct bundle group
d9ae902f52Sean Brightutf8.c: Add UTF-8 validation and utility functions
9022f35f09Sean Brightvector.h: Add AST_VECTOR_SORT()
a678dafac8George JosephCI: Force publishAsteriskDocs to use python2
af70bbb13aJoshua C. Colpwebsocket / pjsip: Increase maximum packet size.
8d15f72721Joshua C. Colppjsip: Include timer patch to prevent cancelling timer 0.
3330764213George JosephUpdate .gitreview defaultbranch to 18
1f5e6805bfAsterisk Development TeamUpdate CHANGES and UPGRADE.txt for 18.0.0
5fbed5af24Ben Fordres_stir_shaken: Add stir_shaken option and general improvements.
e88beedd08George Josephres_pjsip_session: Fix segv in session_on_rx_response
9bd1d686a1George JosephACN: Add tracing to existing code
2d22e34206George JosephACN: res_pjsip endpoint options
d093e44b1eGeorge Josephframe.c: Make debugging easier
955b7b4fdbGeorge JosephScope Trace: Make it easier to trace through synchronous tasks
8d1064eaafGeorge JosephStreams: Add features for Advanced Codec Negotiation
7440fd0397George JosephScope Trace: Add some new tracing macros and an ast_str helper
1274117102Ben Fordres_stir_shaken: Add outbound INVITE support.
f1cfd54976Walter Doekesres_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h"
b9f42a717eGeorge Josephapp_confbridge: Plug ref leak of bridge channel with send_events
3d1bf3c537Kevin HarwellCompiler fixes for gcc 10
559fa0e89cBen Fordcli.c: Fix compiler error.
3927f79cb5Ben Fordres_stir_shaken: Add inbound INVITE support.
1fcb6b1b21Joshua C. Colpbridge_channel: Don't queue unmapped frames.
ca3c22c5f1George JosephScope Tracing: A new facility for tracing scope enter/exit
ec7890d7c6Joshua C. Colpres_sorcery_config: Always reload configuration on errors.
f506cc4896Ben Fordres_stir_shaken: Add unit tests for signing and verification.
e29df34de0Ben Fordres_stir_shaken: Added dialplan function and API call.
44e5dd288bJaco KroonRemove #include <sys/cdefs.h>
1cfd30bd8aJoshua C. Colpres_stir_shaken: Use ast_asprintf for creating file path.
9acf840f7cBen Fordres_stir_shaken: Implemented signature verification.
7baf2c4bf1George Josephapp_voicemail: Add workaround for a gcc 10 issue with -Wrestrict
4ef5ba58f5Alexander TraudBuildSystem: Only if found LibPRI, check its optional parts.
ef580f96e7Alexander TraudBuildSystem: Only if found external PJProject, check its optional parts.
611529fa52Alexander Traudres_stir_shaken: Do not build without OpenSSL.
27de0c9700Alexander Traudres_audiosocket: Avoid Sometimes-uninitialized Warning with Clang.
2b80e5f5daJaco Kroonres_rtp_asterisk: iterate all local addresses looking to populate ICE.
1cf569ba2bJaco Kroonres_pjsip: document legal dtls_verify endpoint options.
52ecbbd014Alexander Traud_pjsua: Build even with Clang.
ee1c7f465bAlexander Traudres_rtp_asterisk: Build without PJProject.
60925c68e8Sean BrightRevert "res_config_odbc: Preserve empty strings returned by the database"
c5f3836bccJaco Kroonmain/backtrace: binutils-2.34 fix.
7ba6d43083George Josephtest_res_pjsip_session_caps: Create unit test
57a457c26cBen Fordres_stir_shaken: Implemented signing of JSON payload.
d32e559e8aJaco Kroonacl: implement a centralized ACL output mechanism for HAs and ACLs.
1b6c58896fJoshua C. Colpchan_sip: Send 403 when ACL fails.
3ed80fc57bJoshua C. ColpCHANGES: Change md file extension to txt.
26713dc88bKevin Harwellast_coredumper: add Asterisk information dump
6f731f153bJaco Kroonnetsock2: compile fixes.
211bb8a79cBen Fordres_stir_shaken: Initial commit and reading private key.
a699e016ddJaco Kroonbuild: enable building with uClibc
f824cd6a13Jaco Kroonbuild: search from newest to oldest for gmime.
82c3939c38Jaco Kroonres_rtp_asterisk: implement ACL mechanism for ICE and STUN addresses.
2ad64e97c0Jaco KroonUpdate main/backtrace.c to deal with changes in binutils 2.34.
49cf84578eSean Brightchan_vpb: Fix 'catching polymorphic type ... by value' error
d68f940f6eSean Brightdns_txt: Add TXT record parsing support
00a7e4b51dGeorge JosephCI: Create generic jenkinsfile
e089779908Rodrigo Ramírez Norambuenares_rtp_asterisk: Add 'rtp show settings' cli command
680e6b9774Walter Doekesapp_queue: Refactor odd placement of if's around say_position
1e1651b4f4Kevin Harwellformat_cap: make function parameters 'const'
0b5c6fddf1Walter Doekessay: Remove unused "plural" option from main/say
5cd7230f3cJaco Kroonaddons/res_config_mysql: silense warnings about printf format errors.
de6919f339Sean Brightast_tls_cert: Allow private key size to be set on command line
8dcdce42a9Sean Brightapp_mixmonitor: Turn on synchronization by default
0f6ee98c3fJoshua C. Colpstasis: Use format specifier for size_t.
1e037ebb97Sean Brightfunc_odbc: Prevent snprintf() truncation warning
a72caa041fGeorge Josephdoc: Fix CHANGES entries to have .txt suffix and update READMEs
1b53d329acJoshua C. Colpres_rtp_asterisk: Don't produce transport-cc if no packets.
b76ab5e5c9George Josephmessage.c: Add option to suppress the Message channel AMI and ARI events
113d05e504Walter Doekeschan_sip: Clarify in sample docs how directmediapermit/-acl should be used
262221f4d9Sean Brightfunc_odbc.conf.sample: Add example lookup
f09cf4da44Sean Brightapp_voicemail: Remove MessageExists and MESSAGE_EXISTS()
5cbf47714aSean Brightapp_voicemail, say: Fix various leading whitespace problems
3bc8b36537Jaco Kroonnetsock2: ast_addressfamily_to_sockaddrsize and ast_sockaddr_from_sockaddr.
00a7432156Kevin Harwellapp_agent_pool: Update XML docs for AgentLogin
19069f7db7Richard Mudgettapp_bridgeaddchan.c: Make BridgeAdd be more like Bridge
0376f2bba9Richard Mudgettfeatures.c: Make Bridge application tolerate unspecified channel.
0d1f3d9bf3Richard Mudgettapp_chanspy.c: Reduce log message level from notice to verbose.
a457947198Richard Mudgettapp_softhangup.c: Reduce unnecessary warning to verbose message.
fc99ac8c9aSean Brightdb: Initialize condition primitive before use
32160cb456Jaco KroonACL: ast_apply_acl_nolog - identical to ast_apply_acl but without logging.
d0b198b330Joshua ColpRevert "PJSIP_CONTACT: add missing argument documentation"
0183e2bc67Sean Brightres_pjsip_registrar.c: Prevent possible buffer overflow with domain aliases
fd823225a6Thomas Arimontchannel.c: Resolve issue with receiving SIP INFO packets for DTMF
366da90f74George JosephCI: Turn off shallow cloning altogether
91c3b5b09dSean Brightmedia_cache.c: Various CLI improvements
48161dfc71Rodrigo Ramírez Norambuenaqueue_log: Add alembic script for generate db table for queue_log
2a6a2800e7George JosephCI: Fix missing script block in jenkinsfiles
4abb54b2e4George JosephCI: Fix missing script block in jenkinsfiles
e8e1314fcbGeorge JosephCI: Increase clone depth and do better cleanup
a5fa0d662eSean Brightres_pjsip_registrar: Fix uninitlized variable warning
f2d5ed54eaAlexei Gradinariserializer: set high/low alert levels on whole pool
bdd785d31cKevin Harwellvarious files - fix some alerts raised by lgtm code analysis
0e3b397812Kevin Harwellres_pjsip_session: initialize pending's topology to endpoint's
8a1f30af04Corey Farrellcore: Improve MALLOC_DEBUG for frames.
d71d0f9489George JosephExternalMedia: Change return object from ExternalMedia to Channel
6e907ae5d4Joshua Colpres_rtp_asterisk: Remove a log message that slipped in.
a60d2e905cJoshua Colptest_res_rtp: Enable FIR and REMB nominal tests.
b27a5183daChris Savinovichtest_taskprocessor.c: Fix test failure on Ubuntu
c0efe19cecKevin Harwellserializer: move/add asterisk serializer pool functionality
2970a13fb8Kevin Harwellres_pjsip/res_pjsip_mwi: use centralized serializer pools
068ed2c626Alexei Gradinarires_pjsip_pubsub: add endpoint to some warning
ba64d68273Jonathan Rosebasic-pbx: Bring forward queue configuration from 13
4c3655ecfdBen Fordtaskprocessor.c: Added "like" support to 'core show taskprocessors'
966488ab52Sean Brightres_musiconhold: Add new 'playlist' mode
f7045cefd9Corey Farrellstasis_state: Create internal stasis_state_proxy object.
67ba62f4e6Kevin Harwellres_pjsip_pubsub: change warning to debug
4de1e6d0e6Ben Fordtaskprocessor.c: Add CLI commands to reset taskprocessor stats.
725e991fafCorey Farrellcore: Add AO2_ALLOC_OPT_NO_REF_DEBUG option.
e82f2f6e82George Josephastmm.c: Display backtrace with memory show allocations
a4142c8437Corey Farrellcore: Fix ABI mismatch of ao2_global_obj.
ca608d2575Corey Farrellstasis: refcounter.py can incorrectly report skewed objects.
3dfbc05c53Corey Farrellstasis: Fix leaks
863fe2225fCorey Farrellapp_voicemail: Fix module unload leak.
723b695ce5Ben Fordres_rtp_asterisk.c: Send RTCP as compound packets.
0e56643d9fBen Fordres_rtp: Add unit tests for RTCP stats.
2ae1a22e0eGeorge JosephARI: External Media
5fb9b23105George Josephchan_sip: Update links referenced in deprecation notice
ed757cc7bbChris-Savinovichtest_utils.c: Skip test adsi_loaded_test if module not loaded.
1d06a1efb3Igor Goncharovskychan_unistim: Fix code, causing all incoming DTMF sent back to asterisk
649003821eIgor Goncharovskychan_unistim: Fix RTP port byte order for big-endian arch
3ef52b0b17Alexei GradinariFix misname 'res_external_mwi' to 'res_mwi_external' in comments.
19045db392George Josephchan_rtp: Accept hostname as well as ip address as destination
9e015713ccGeorge Josephdns_core: Create new API ast_dns_resolve_ipv6_and_ipv4
8da4e28a81George Josephres_ari.c: Prefer exact handler match over wildcard
64906c4c9bSean Brightaudiohook.c: Substitute silence for unavailable audio frames
446bac733dGeorge JosephCI: Escape backslashes in printenv/sort/tr
be6130607dGeorge JosephCI: Add "throttle" label and "skip_gate" capability
c01dd2a41aGeorge JosephCI: Make node labels job-specific
9d07d5a6d6Sean Brightapp_voicemail: Remove extra menuselect build options
1f8ae708a0Sean Brightres_musiconhold: Use a vector instead of custom array allocation
5f66fb5139Sean Brightmanager: Send fewer packets
5e6e1175d5Asterisk Development TeamUpdate CHANGES and UPGRADE.txt for 17.0.0
8d10028b98George JosephUpdate master for Asterisk 18
7ce9ee7f2eSean Brightres_musiconhold: Use ast_pipe_nonblock() wrapper
8e44d823c1George Josephloader.c: Fix possible SEGV when a module fails to register

Diffstat Results

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This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.

asterisk-17.0.0-summary.html                                                                  | 1265 --
+asterisk-17.0.0-summary.txt                                                                   | 2973 ----
+b/.gitreview                                                                                  |    2
+b/.version                                                                                    |    2
+b/CHANGES                                                                                     |  400
+b/ChangeLog                                                                                   | 6317 +++++++++-
+b/Makefile                                                                                    |    8
+b/Makefile.rules                                                                              |   19
+b/README-SERIOUSLY.bestpractices.md                                                           |    4
+b/UPGRADE.txt                                                                                 |  120
+b/addons/cdr_mysql.c                                                                          |   18
+b/addons/chan_mobile.c                                                                        |   63
+b/addons/chan_ooh323.c                                                                        |    7
+b/addons/ooh323c/src/decode.c                                                                 |   10
+b/addons/ooh323c/src/ooSocket.c                                                               |    2
+b/addons/ooh323c/src/oochannels.c                                                             |    2
+b/addons/res_config_mysql.c                                                                   |   16
+b/apps/app_agent_pool.c                                                                       |    4
+b/apps/app_amd.c                                                                              |   24
+b/apps/app_audiosocket.c                                                                      |  240
+b/apps/app_bridgeaddchan.c                                                                    |   62
+b/apps/app_cdr.c                                                                              |    8
+b/apps/app_chanisavail.c                                                                      |  137
+b/apps/app_chanspy.c                                                                          |    3
+b/apps/app_confbridge.c                                                                       |    9
+b/apps/app_dahdiras.c                                                                         |    6
+b/apps/app_dial.c                                                                             |  108
+b/apps/app_dictate.c                                                                          |    4
+b/apps/app_fax.c                                                                              |    2
+b/apps/app_followme.c                                                                         |   10
+b/apps/app_getcpeid.c                                                                         |    1
+b/apps/app_meetme.c                                                                           |    2
+b/apps/app_minivm.c                                                                           |    3
+b/apps/app_mixmonitor.c                                                                       |   51
+b/apps/app_osplookup.c                                                                        |   13
+b/apps/app_page.c                                                                             |   30
+b/apps/app_playback.c                                                                         |    9
+b/apps/app_queue.c                                                                            |  201
+b/apps/app_readexten.c                                                                        |    3
+b/apps/app_record.c                                                                           |    3
+b/apps/app_senddtmf.c                                                                         |   13
+b/apps/app_softhangup.c                                                                       |    2
+b/apps/app_stack.c                                                                            |   11
+b/apps/app_stream_echo.c                                                                      |    2
+b/apps/app_voicemail.c                                                                        |  839 -
+b/apps/confbridge/conf_config_parser.c                                                        |   30
+b/apps/confbridge/confbridge_manager.c                                                        |    2
+b/apps/confbridge/include/confbridge.h                                                        |    2
+b/asterisk-18.0.0-rc2-summary.html                                                            |   15
+b/asterisk-18.0.0-rc2-summary.txt                                                             |   83
+b/bridges/bridge_native_rtp.c                                                                 |  173
+b/bridges/bridge_simple.c                                                                     |  200
+b/bridges/bridge_softmix.c                                                                    |  281
+b/cdr/cdr_odbc.c                                                                              |    2
+b/cdr/cdr_pgsql.c                                                                             |    2
+b/cel/cel_pgsql.c                                                                             |    2
+b/channels/Makefile                                                                           |    2
+b/channels/chan_audiosocket.c                                                                 |  302
+b/channels/chan_dahdi.c                                                                       |   39
+b/channels/chan_dahdi.h                                                                       |   18
+b/channels/chan_iax2.c                                                                        |   16
+b/channels/chan_motif.c                                                                       |    9
+b/channels/chan_pjsip.c                                                                       |  314
+b/channels/chan_rtp.c                                                                         |   19
+b/channels/chan_sip.c                                                                         |  268
+b/channels/chan_unistim.c                                                                     |  178
+b/channels/chan_vpb.cc                                                                        |    2
+b/channels/iax2/parser.c                                                                      |   18
+b/channels/pjsip/cli_commands.c                                                               |   13
+b/channels/pjsip/dialplan_functions.c                                                         |   84
+b/channels/pjsip/include/dialplan_functions.h                                                 |   25
+b/channels/sig_pri.c                                                                          |   25
+b/channels/sip/include/sip.h                                                                  |    1
+b/codecs/Makefile                                                                             |    3
+b/codecs/ex_alaw.h                                                                            |    5
+b/codecs/ex_g722.h                                                                            |    5
+b/codecs/ex_ulaw.h                                                                            |    5
+b/codecs/speex/arch.h                                                                         |   13
+b/codecs/speex/fixed_generic.h                                                                |    4
+b/codecs/speex/resample.c                                                                     |  332
+b/codecs/speex/speex_resampler.h                                                              |    4
+b/configs/basic-pbx/extensions.conf                                                           |   14
+b/configs/basic-pbx/modules.conf                                                              |    1
+b/configs/basic-pbx/queues.conf                                                               |   19
+b/configs/samples/asterisk.conf.sample                                                        |    6
+b/configs/samples/confbridge.conf.sample                                                      |    7
+b/configs/samples/extconfig.conf.sample                                                       |    1
+b/configs/samples/func_odbc.conf.sample                                                       |   19
+b/configs/samples/http.conf.sample                                                            |   10
+b/configs/samples/indications.conf.sample                                                     |   11
+b/configs/samples/logger.conf.sample                                                          |   15
+b/configs/samples/musiconhold.conf.sample                                                     |   23
+b/configs/samples/pjsip.conf.sample                                                           |   79
+b/configs/samples/queues.conf.sample                                                          |    2
+b/configs/samples/rtp.conf.sample                                                             |   30
+b/configs/samples/sip.conf.sample                                                             |    4
+b/configs/samples/stir_shaken.conf.sample                                                     |   61
+b/configure                                                                                   |  666 -
+b/configure.ac                                                                                |  196
+b/contrib/ast-db-manage/README.md                                                             |    1
+b/contrib/ast-db-manage/config/versions/61797b9fced6_add_stir_shaken.py                       |   31
+b/contrib/ast-db-manage/config/versions/79290b511e4b_pjsip_add_disable_rport.py               |   39
+b/contrib/ast-db-manage/config/versions/b80485ff4dd0_add_pjsip_endpoint_acn_options.py        |   29
+b/contrib/ast-db-manage/config/versions/fbb7766f17bc_add_playlist_to_moh.py                   |   54
+b/contrib/ast-db-manage/queue_log.ini.sample                                                  |   58
+b/contrib/ast-db-manage/queue_log/env.py                                                      |    1
+b/contrib/ast-db-manage/queue_log/script.py.mako                                              |   24
+b/contrib/ast-db-manage/queue_log/versions/4105ee839f58_create_queue_log_table.py             |   38
+b/contrib/realtime/mysql/mysql_config.sql                                                     |   39
+b/contrib/realtime/postgresql/postgresql_config.sql                                           |   45
+b/contrib/scripts/ast_coredumper                                                              |  421
+b/contrib/scripts/ast_tls_cert                                                                |    8
+b/contrib/scripts/install_prereq                                                              |   30
+b/contrib/scripts/sip_nat_settings                                                            |   19
+b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py                                                |    2
+b/contrib/valgrind.supp                                                                       |   14
+b/doc/CHANGES-staging/README.md                                                               |    8
+b/doc/CHANGES-staging/hide_messaging_ami_events                                               |   11
+b/doc/UPGRADE-staging/README.md                                                               |    7
+b/doc/appdocsxml.dtd                                                                          |    2
+b/formats/format_g726.c                                                                       |   16
+b/formats/msgsm.h                                                                             |    4
+b/funcs/func_aes.c                                                                            |    4
+b/funcs/func_channel.c                                                                        |    8
+b/funcs/func_curl.c                                                                           |   49
+b/funcs/func_env.c                                                                            |    5
+b/funcs/func_jitterbuffer.c                                                                   |   19
+b/funcs/func_odbc.c                                                                           |   26
+b/funcs/func_pjsip_contact.c                                                                  |    6
+b/funcs/func_volume.c                                                                         |   12
+b/include/asterisk/abstract_jb.h                                                              |    4
+b/include/asterisk/acl.h                                                                      |   69
+b/include/asterisk/app.h                                                                      |    7
+b/include/asterisk/ari.h                                                                      |    2
+b/include/asterisk/astobj2.h                                                                  |   61
+b/include/asterisk/audiohook.h                                                                |    2
+b/include/asterisk/autoconfig.h.in                                                            |    9
+b/include/asterisk/bridge.h                                                                   |   21
+b/include/asterisk/bridge_features.h                                                          |    4
+b/include/asterisk/calendar.h                                                                 |    4
+b/include/asterisk/channel.h                                                                  |   87
+b/include/asterisk/channel_internal.h                                                         |    5
+b/include/asterisk/config.h                                                                   |   18
+b/include/asterisk/config_options.h                                                           |    2
+b/include/asterisk/conversions.h                                                              |   54
+b/include/asterisk/core_unreal.h                                                              |   19
+b/include/asterisk/dns_core.h                                                                 |   22
+b/include/asterisk/dns_internal.h                                                             |   31
+b/include/asterisk/dns_txt.h                                                                  |   64
+b/include/asterisk/format_cache.h                                                             |    5
+b/include/asterisk/format_cap.h                                                               |   15
+b/include/asterisk/frame.h                                                                    |   40
+b/include/asterisk/http_websocket.h                                                           |   14
+b/include/asterisk/iostream.h                                                                 |   14
+b/include/asterisk/logger.h                                                                   |  426
+b/include/asterisk/manager.h                                                                  |    2
+b/include/asterisk/max_forwards.h                                                             |    1
+b/include/asterisk/message.h                                                                  |   31
+b/include/asterisk/mixmonitor.h                                                               |    5
+b/include/asterisk/netsock2.h                                                                 |   42
+b/include/asterisk/options.h                                                                  |    7
+b/include/asterisk/parking.h                                                                  |    5
+b/include/asterisk/res_audiosocket.h                                                          |   87
+b/include/asterisk/res_fax.h                                                                  |    3
+b/include/asterisk/res_pjsip.h                                                                |  106
+b/include/asterisk/res_pjsip_presence_xml.h                                                   |    5
+b/include/asterisk/res_pjsip_pubsub.h                                                         |   23
+b/include/asterisk/res_pjsip_session.h                                                        |   27
+b/include/asterisk/res_pjsip_session_caps.h                                                   |   82
+b/include/asterisk/res_stir_shaken.h                                                          |  117
+b/include/asterisk/rtp_engine.h                                                               |  111
+b/include/asterisk/say.h                                                                      |    4
+b/include/asterisk/serializer.h                                                               |   85
+b/include/asterisk/slin.h                                                                     |    5
+b/include/asterisk/sorcery.h                                                                  |   27
+b/include/asterisk/stasis.h                                                                   |    3
+b/include/asterisk/stasis_app.h                                                               |   10
+b/include/asterisk/stasis_channels.h                                                          |    8
+b/include/asterisk/stream.h                                                                   |  463
+b/include/asterisk/strings.h                                                                  |   53
+b/include/asterisk/taskprocessor.h                                                            |    9
+b/include/asterisk/utf8.h                                                                     |  188
+b/include/asterisk/utils.h                                                                    |   59
+b/include/asterisk/vector.h                                                                   |   50
+b/main/Makefile                                                                               |    1
+b/main/abstract_jb.c                                                                          |  178
+b/main/acl.c                                                                                  |  105
+b/main/app.c                                                                                  |   26
+b/main/ast_expr2.c                                                                            |    1
+b/main/ast_expr2.y                                                                            |    1
+b/main/asterisk.c                                                                             |   41
+b/main/asterisk.exports.in                                                                    |    1
+b/main/astmm.c                                                                                |   23
+b/main/astobj2.c                                                                              |   88
+b/main/astobj2_container.c                                                                    |   24
+b/main/astobj2_global.c                                                                       |   97
+b/main/astobj2_hash.c                                                                         |   21
+b/main/astobj2_rbtree.c                                                                       |   13
+b/main/audiohook.c                                                                            |   20
+b/main/backtrace.c                                                                            |    9
+b/main/bridge.c                                                                               |   15
+b/main/bridge_channel.c                                                                       |   62
+b/main/cdr.c                                                                                  |   15
+b/main/channel.c                                                                              |  204
+b/main/channel_internal_api.c                                                                 |   12
+b/main/cli.c                                                                                  |  261
+b/main/codec_builtin.c                                                                        |    8
+b/main/config.c                                                                               |   28
+b/main/conversions.c                                                                          |   51
+b/main/core_local.c                                                                           |  112
+b/main/core_unreal.c                                                                          |  141
+b/main/data_buffer.c                                                                          |    2
+b/main/db.c                                                                                   |    3
+b/main/dial.c                                                                                 |   14
+b/main/dns_core.c                                                                             |   75
+b/main/dns_srv.c                                                                              |    6
+b/main/dns_txt.c                                                                              |  127
+b/main/enum.c                                                                                 |  104
+b/main/event.c                                                                                |   17
+b/main/features.c                                                                             |   38
+b/main/file.c                                                                                 |   46
+b/main/format_cache.c                                                                         |    8
+b/main/format_cap.c                                                                           |   24
+b/main/frame.c                                                                                |  160
+b/main/http.c                                                                                 |   62
+b/main/indications.c                                                                          |    6
+b/main/iostream.c                                                                             |   14
+b/main/logger.c                                                                               |  150
+b/main/manager.c                                                                              |  210
+b/main/media_cache.c                                                                          |   47
+b/main/message.c                                                                              |   55
+b/main/named_acl.c                                                                            |    9
+b/main/options.c                                                                              |   11
+b/main/pbx.c                                                                                  |   35
+b/main/pbx_variables.c                                                                        |   31
+b/main/rtp_engine.c                                                                           |  137
+b/main/say.c                                                                                  |  968 -
+b/main/serializer.c                                                                           |  189
+b/main/sorcery.c                                                                              |   46
+b/main/stasis.c                                                                               |   53
+b/main/stasis_cache.c                                                                         |   10
+b/main/stasis_channels.c                                                                      |    7
+b/main/stasis_state.c                                                                         |  298
+b/main/strcompat.c                                                                            |   94
+b/main/stream.c                                                                               |  432
+b/main/taskprocessor.c                                                                        |  219
+b/main/tcptls.c                                                                               |   33
+b/main/translate.c                                                                            |    8
+b/main/utf8.c                                                                                 |  380
+b/main/utils.c                                                                                |   50
+b/makeopts.in                                                                                 |    6
+b/menuselect/configure                                                                        |   14
+b/menuselect/menuselect.c                                                                     |   14
+b/menuselect/menuselect.h                                                                     |    2
+b/pbx/pbx_dundi.c                                                                             |   21
+b/res/Makefile                                                                                |    2
+b/res/ari/ari_model_validators.c                                                              |   59
+b/res/ari/ari_model_validators.h                                                              |   23
+b/res/ari/config.c                                                                            |   10
+b/res/ari/resource_bridges.c                                                                  |    1
+b/res/ari/resource_bridges.h                                                                  |    2
+b/res/ari/resource_channels.c                                                                 |  245
+b/res/ari/resource_channels.h                                                                 |   46
+b/res/ari/resource_events.c                                                                   |   10
+b/res/parking/parking_bridge.c                                                                |   36
+b/res/parking/parking_bridge_features.c                                                       |   14
+b/res/parking/res_parking.h                                                                   |    5
+b/res/res_agi.c                                                                               |   20
+b/res/res_ari_bridges.c                                                                       |    7
+b/res/res_ari_channels.c                                                                      |  147
+b/res/res_ari_events.c                                                                        |    2
+b/res/res_audiosocket.c                                                                       |  345
+b/res/res_audiosocket.exports.in                                                              |    4
+b/res/res_calendar_ews.c                                                                      |    1
+b/res/res_calendar_exchange.c                                                                 |    1
+b/res/res_calendar_icalendar.c                                                                |    1
+b/res/res_config_curl.c                                                                       |    7
+b/res/res_config_pgsql.c                                                                      |    2
+b/res/res_corosync.c                                                                          |  564
+b/res/res_fax.c                                                                               |   37
+b/res/res_http_websocket.c                                                                    |   97
+b/res/res_musiconhold.c                                                                       |  472
+b/res/res_phoneprov.c                                                                         |    6
+b/res/res_pjsip.c                                                                             |  492
+b/res/res_pjsip/config_system.c                                                               |    8
+b/res/res_pjsip/config_transport.c                                                            |   17
+b/res/res_pjsip/location.c                                                                    |   25
+b/res/res_pjsip/pjsip_configuration.c                                                         |  209
+b/res/res_pjsip/pjsip_message_filter.c                                                        |   40
+b/res/res_pjsip/pjsip_options.c                                                               |    7
+b/res/res_pjsip/pjsip_resolver.c                                                              |   30
+b/res/res_pjsip_acl.c                                                                         |   20
+b/res/res_pjsip_caller_id.c                                                                   |    3
+b/res/res_pjsip_config_wizard.c                                                               |    7
+b/res/res_pjsip_dialog_info_body_generator.c                                                  |   80
+b/res/res_pjsip_diversion.c                                                                   |    7
+b/res/res_pjsip_dlg_options.c                                                                 |    3
+b/res/res_pjsip_dtmf_info.c                                                                   |    1
+b/res/res_pjsip_empty_info.c                                                                  |    1
+b/res/res_pjsip_endpoint_identifier_ip.c                                                      |  102
+b/res/res_pjsip_history.c                                                                     |    4
+b/res/res_pjsip_logger.c                                                                      |  451
+b/res/res_pjsip_messaging.c                                                                   |   58
+b/res/res_pjsip_mwi.c                                                                         |  339
+b/res/res_pjsip_nat.c                                                                         |  110
+b/res/res_pjsip_notify.c                                                                      |   22
+b/res/res_pjsip_one_touch_record_info.c                                                       |    1
+b/res/res_pjsip_outbound_registration.c                                                       |   66
+b/res/res_pjsip_path.c                                                                        |    1
+b/res/res_pjsip_pubsub.c                                                                      |  148
+b/res/res_pjsip_refer.c                                                                       |    8
+b/res/res_pjsip_registrar.c                                                                   |   80
+b/res/res_pjsip_rfc3326.c                                                                     |    1
+b/res/res_pjsip_sdp_rtp.c                                                                     |  283
+b/res/res_pjsip_session.c                                                                     |  667 -
+b/res/res_pjsip_session.exports.in                                                            |    1
+b/res/res_pjsip_session/pjsip_session_caps.c                                                  |  164
+b/res/res_pjsip_stir_shaken.c                                                                 |  330
+b/res/res_pjsip_t38.c                                                                         |   40
+b/res/res_pjsip_transport_websocket.c                                                         |    4
+b/res/res_realtime.c                                                                          |   56
+b/res/res_resolver_unbound.c                                                                  |    6
+b/res/res_rtp_asterisk.c                                                                      | 1100 +
+b/res/res_sorcery_config.c                                                                    |   14
+b/res/res_sorcery_memory_cache.c                                                              |   18
+b/res/res_speech.c                                                                            |    7
+b/res/res_srtp.c                                                                              |    8
+b/res/res_stasis.c                                                                            |   57
+b/res/res_stasis_playback.c                                                                   |    4
+b/res/res_statsd.c                                                                            |   35
+b/res/res_stir_shaken.c                                                                       | 1663 ++
+b/res/res_stir_shaken.exports.in                                                              |    6
+b/res/res_stir_shaken/certificate.c                                                           |  388
+b/res/res_stir_shaken/certificate.h                                                           |  119
+b/res/res_stir_shaken/curl.c                                                                  |  199
+b/res/res_stir_shaken/curl.h                                                                  |   73
+b/res/res_stir_shaken/general.c                                                               |  286
+b/res/res_stir_shaken/general.h                                                               |  111
+b/res/res_stir_shaken/stir_shaken.c                                                           |  119
+b/res/res_stir_shaken/stir_shaken.h                                                           |   55
+b/res/res_stir_shaken/store.c                                                                 |  202
+b/res/res_stir_shaken/store.h                                                                 |   37
+b/res/stasis/app.c                                                                            |   15
+b/res/stasis/command.c                                                                        |    2
+b/res/stasis/control.c                                                                        |   14
+b/res/stasis/messaging.c                                                                      |   11
+b/res/stasis/stasis_bridge.c                                                                  |   16
+b/res/stasis/stasis_bridge.h                                                                  |    3
+b/rest-api-templates/make_ari_stubs.py                                                        |    2
+b/rest-api-templates/res_ari_resource.c.mustache                                              |    2
+b/rest-api/api-docs/bridges.json                                                              |    9
+b/rest-api/api-docs/channels.json                                                             |  143
+b/rest-api/api-docs/endpoints.json                                                            |   20
+b/rest-api/resources.json                                                                     |    2
+b/tests/CI/buildAsterisk.sh                                                                   |    2
+b/tests/CI/gates.jenkinsfile                                                                  |   12
+b/tests/CI/periodics-daily.jenkinsfile                                                        |   11
+b/tests/CI/publishAsteriskDocs.sh                                                             |    4
+b/tests/CI/ref_debug.jenkinsfile                                                              |    9
+b/tests/CI/unittests.jenkinsfile                                                              |    9
+b/tests/CI/universal-asterisk-nongerrit.jenkinsfile                                           |  452
+b/tests/test_conversions.c                                                                    |  153
+b/tests/test_data_buffer.c                                                                    |    2
+b/tests/test_locale.c                                                                         |   12
+b/tests/test_res_pjsip_session_caps.c                                                         |  176
+b/tests/test_res_rtp.c                                                                        |  516
+b/tests/test_scope_trace.c                                                                    |  126
+b/tests/test_stasis.c                                                                         |    2
+b/tests/test_stream.c                                                                         |    2
+b/tests/test_strings.c                                                                        |   39
+b/tests/test_taskprocessor.c                                                                  |   78
+b/tests/test_utils.c                                                                          |   11
+b/third-party/pjproject/Makefile                                                              |    3
+b/third-party/pjproject/configure.m4                                                          |  199
+b/third-party/pjproject/patches/0011-sip_inv_patch.patch                                      |   39
+b/third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch                               |   39
+b/third-party/pjproject/patches/0050-fix-race-parallel-build.patch                            |   72
+b/third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch             |   28
+b/third-party/pjproject/patches/config_site.h                                                 |    4
+b/third-party/pjproject/pjproject-2.10.tar.bz2.md5                                            |    2
+b/third-party/versions.mak                                                                    |    2
+b/utils/Makefile                                                                              |    6
+b/utils/astman.c                                                                              |    2
+b/utils/db1-ast/hash/ndbm.c                                                                   |    3
+b/utils/db1-ast/include/db.h                                                                  |    1
+b/utils/extconf.c                                                                             |   10
+b/utils/muted.c                                                                               |   11
+third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch |  157
+third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch                      |   53
+third-party/pjproject/patches/0030-ssl-regression-fix.patch                                   |  105
+third-party/pjproject/patches/0031-transport-regression-fix.patch                             |  187
+third-party/pjproject/pjproject-2.9.tar.bz2.md5                                               |    2
+392 files changed, 29541 insertions(+), 9926 deletions(-)

\ No newline at end of file diff --git a/asterisk-18.0.0-summary.txt b/asterisk-18.0.0-summary.txt new file mode 100644 index 0000000000..53617db7cb --- /dev/null +++ b/asterisk-18.0.0-summary.txt @@ -0,0 +1,2873 @@ + Release Summary + + asterisk-18.0.0 + + Date: 2020-10-19 + + + + ---------------------------------------------------------------------- + + Table of Contents + + 1. Summary + 2. Contributors + 3. Closed Issues + 4. Open Issues + 5. Other Changes + 6. Diffstat + + ---------------------------------------------------------------------- + + Summary + + [Back to Top] + + This is the first release of a major new version of Asterisk. For a list + of new features that have been included with this release, please see the + CHANGES file inside the source package. Since this is a new major release, + users are encouraged to do extended testing before upgrading to this + version in a production environment. + + The data in this summary reflects changes that have been made since the + previous release, asterisk-17.0.0. + + ---------------------------------------------------------------------- + + Contributors + + [Back to Top] + + This table lists the people who have submitted code, those that have + tested patches, as well as those that reported issues on the issue tracker + that were resolved in this release. For coders, the number is how many of + their patches (of any size) were committed into this release. For testers, + the number is the number of times their name was listed as assisting with + testing a patch. Finally, for reporters, the number is the number of + issues that they reported that were affected by commits that went into + this release. + + Coders Testers Reporters + 70 Sean Bright 1 tests/test_utils.c. 26 Joshua C. Colp + 54 George Joseph 23 Alexander Traud + 47 Joshua C. Colp 13 Kevin Harwell + 31 Kevin Harwell 9 Ross Beer + 26 Alexander Traud 8 sungtae kim + 17 Ben Ford 8 nappsoft + 16 Joshua Colp 7 Walter Doekes + 16 Jaco Kroon 7 Torrey Searle + 11 Walter Doekes 6 Frederic LE FOLL + 9 sungtae kim 5 Guido Falsi + 9 Alexander Traud 5 cmaj + 8 Torrey Searle 5 George Joseph + 8 Corey Farrell 4 Jaco Kroon + 7 Richard Mudgett 4 Jean Aunis - Prescom + 6 Frederic LE FOLL 4 Pascal Cadotte Michaud + 6 Pirmin Walthert 4 Ross Beer + 6 Guido Falsi 3 Joshua Elson + 5 Asterisk Development 3 Sean Bright + Team 3 Salah Ahmed + 5 Alexei Gradinari 3 Nickolay V. Shmyrev + 3 Pascal Cadotte Michaud 3 lvl + 3 Igor Goncharovsky 3 Alexei Gradinari + 3 Jean Aunis 2 Stas Kobzar + 3 Nickolay Shmyrev 2 Michael Neuhauser + 3 lvl 2 Ruddy G + 2 Rodrigo RamÃrez 2 Joeran Vinzens + Norambuena 2 Timothy Vanderaerden + 2 Andrew Siplas 2 Sébastien Duthil + 2 Salah Ahmed 2 Peter Sokolov + 2 Michael Neuhauser 2 Joseph Ades + 2 cmaj 2 Gregory Massel + 1 Kevin Reeves 2 Andrew Siplas + 1 Michael Goryainov 2 Jared Smith + 1 Università di Bologna - 2 Jonathan Harris + CESIA VoIP 2 Michael Neuhauser + 1 Chris Savinovich 1 Ramarajan + 1 Nathan Bruning 1 Andrey V. T. + 1 Paulo Vicentini 1 tootai + 1 Peter Turczak 1 Martin Tomec + 1 Sungtae Kim 1 AvayaXAsterisk + 1 Dan Cropp 1 Joshua C. Colp + 1 Jared Smith 1 Etienne Lessard + 1 Stas Kobzar 1 Benjamin Keith Ford + 1 Daniel Heckl 1 Yoooooo Ha + 1 Dennis Buteyn 1 kevin@phoneburner.com + 1 Bernard Merindol 1 Juan Martin + 1 Jonathan Rose 1 Sylvain Afchain + 1 Kfir Itzhak 1 Speed Dial Dave + 1 Roger James 1 Andrew Yager + 1 Sebastian Kemper 1 Jean-Denis Girard + 1 Christoph Moench-Tegeder 1 Marian Piater + 1 Boris P. Korzun 1 Bernard Merindol + 1 Evandro César Arruda 1 Martin Zeh + 1 Moises Silva 1 Corey Farrell + 1 Chris-Savinovich 1 Dan Cropp + 1 Michael Cargile 1 Moises Silva + 1 Sylvain Afchain 1 Alexey Vasilyev + 1 Florian Floimair 1 Thomas Johnson + 1 Nicholas John Koch 1 Seán C. McCord + 1 Peter Sokolov (License 1 Dirk Wendland + #7070) 1 Bryan Nelson + 1 Martin Tomec 1 Sam Banks + 1 Thomas Arimont (license 1 Misha Vodsedalek + 5525) 1 Nicholas John Koch + 1 Seán C McCord 1 Richard Kenner + 1 Patrick Verzele 1 EDV O-TON + 1 snuffy 1 Byron Clark + 1 Sebastien Duthil 1 Christoph + 1 Jason Hord (license Moench-Tegeder + 6978) 1 sstream + 1 Dmitriy Serov + 1 Alex + 1 candrews + 1 Sébastien Duthil + 1 Robert Sutton + 1 Evandro César Arruda + 1 Paul Brooks + 1 Yury Kirsanov + 1 Jason Hord + 1 Michael Cargile + 1 Kevin Flyn + 1 Shlomi Gutman + 1 George Joseph + 1 Frank Matano + 1 Cédric Bassaget + 1 Dan Jenkins + 1 Jim Van Meggelen + 1 Patrick Wakano + 1 Jeremiah Gadd + 1 Michael + 1 Daniel Heckl + 1 Boris P. Korzun + 1 Kilburn + 1 Bernhard Schmidt + 1 Alexander Traud + 1 Joeran Vinzens + 1 Dennis + 1 Vitold + 1 Anton Satskiy + 1 Kevin Flyn + 1 David Cunningham + 1 Jim Van Meggelen + 1 Vitold + 1 Florian Floimair + 1 Robert Sutton + 1 Daniel + 1 Dan Jenkins + 1 Ove Aursand + 1 Dmitry Wagin + 1 Robin Leffmann + 1 Mitch Claborn + 1 Jonathan Hunter + 1 Joshua Roys + 1 Olivier Krief + 1 Paul Brooks + 1 Maciej Michno + 1 Kevin Reeves + 1 Niklas Larsson + 1 Bernhard Schmidt + 1 Christoph + Moench-Tegeder + 1 Maciej Michno + 1 Stas Kobzar + 1 Cedric BASSAGET + 1 EDV O-TON + 1 Ted G + 1 Frank Matano + 1 Yury Kirsanov + 1 Anton Satskiy + 1 David M. Lee + 1 Patrick Wakano + 1 Michael Goryainov + 1 Niklas Larsson + 1 Sebastian Kemper + 1 Francois Blackburn + 1 Università di Bologna + - CESIA VoIP + 1 Richard Kenner + 1 Niksa Baldun + 1 Ian Jones + 1 Jean-Denis Girard + 1 Dmitriy Serov + 1 Peter Turczak + 1 Roger James + 1 Paulo Vicentini + 1 Ted G + 1 Martin Zeh + 1 Università di Bologna + - CESIA VoIP + 1 Marin Odrljin + 1 Jonas Swiatek + 1 Eliel Sardañons + 1 AvayaXAsterisk + 1 Dirk Wendland + 1 Joshua Roys + 1 Mark + 1 Dan Cropp + 1 Jonathan Harris + 1 Matt Addison + 1 Leandro Dardini + 1 alex + 1 Chris Savinovich + 1 xrobau + 1 David Lee + 1 Nicholas John Koch + 1 Peter Sokolov + 1 Eliel Sardañons + 1 Sean Bright + 1 Aheliotech + 1 Bill Kervaski + 1 Cyril Ramière + 1 Jørgen H + 1 Niksa Baldun + 1 Kfir Itzhak + + ---------------------------------------------------------------------- + + Closed Issues + + [Back to Top] + + This is a list of all issues from the issue tracker that were closed by + changes that went into this release. + + Security + + Category: Channels/chan_sip/General + + ASTERISK-28589: chan_sip: Depending on configuration an INVITE can alter + Addr of a peer + Reported by: Andrey V. T. + * [4a1cadeadb] Ben Ford -- chan_sip.c: Prevent address change on + unauthenticated SIP request. + + Category: Core/ManagerInterface + + ASTERISK-28580: Bypass SYSTEM write permission in manager action allows + system commands execution + Reported by: Eliel Sardañons + * [7e3a6e158f] George Joseph -- manager.c: Prevent the Originate action + from running the Originate app + + Category: Resources/res_pjsip_t38 + + ASTERISK-28495: res_pjsip_t38: 200 OK with SDP answer with declined stream + causes crash + Reported by: Alexei Gradinari + * [18f5f5fc99] Alexei Gradinari -- AST-2019-004 - res_pjsip_t38.c: Add + NULL checks before using session media + + New Feature + + Category: Applications/app_senddtmf + + ASTERISK-28614: app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead + of only "sending" + Reported by: lvl + * [772b59034f] lvl -- app_senddtmf: Add receive mode to AMI Action + PlayDTMF + + Category: Core/General + + ASTERISK-6863: [patch] allow Asterisk to set high ToS bits as non-root on + Linux + Reported by: Matt Addison + * [a107e85b2e] Alexander Traud -- install_prereq: Add libcap for high + bits in DiffServ/ToS. + + Category: Core/Jitterbuffer + + ASTERISK-28533: func_jitterbuffer: Add support for video synchronization + Reported by: Joshua C. Colp + * [7298a785ad] Joshua Colp -- func_jitterbuffer: Add audio/video sync + support. + + Category: Functions/func_curl + + ASTERISK-17491: CURLOPT() needs a "followlocation" parameter / "maxredirs" + doesn't do anything + Reported by: candrews + * [0c2bf1664c] Sean Bright -- func_curl: Add 'followlocation' option to + CURLOPT() + ASTERISK-28613: func_curl: CURLOPT cannot set Content-Type header + Reported by: Martin Tomec + * [d257a0898e] Martin Tomec -- func_curl.c: Support custom http headers + + Category: Resources/res_musiconhold + + ASTERISK-17808: [patch] Unregister a realtime moh class + Reported by: Byron Clark + * [cf364cd007] sungtae kim -- res_musiconhold: Added unregister realtime + moh class + + Category: Resources/res_pjsip_endpoint_identifier_ip + + ASTERISK-28639: res_pjsip_endpoint_identifier_ip: Add ability to match on + source port + Reported by: Sean Bright + * [312abaa1fe] Sean Bright -- res_pjsip_endpoint_identifier_ip.c: Add + port matching support + + Category: pjproject/pjsip + + ASTERISK-28489: Channel variable SIPFROMDOMAIN for chan_pjsip to setup + From header URI domain + Reported by: Stas Kobzar + * [c7270dca81] Stas Kobzar -- res_pjsip: Channel variable SIPFROMDOMAIN + + Bug + + Category: .Release/Targets + + ASTERISK-28488: pjsip mwi: n+1 sip notify's sent on re-register + Reported by: Chris Savinovich + * [172e183b9d] Kevin Harwell -- res_pjsip_mwi: add better handling of + solicited vs unsolicited subscriptions + + Category: Applications/General + + ASTERISK-28954: StreamEcho() only returns 1 active stream + Reported by: Bill Kervaski + * [00a52b4752] Joshua C. Colp -- app_stream_echo: Fix state of added + streams. + ASTERISK-16676: DAHDIRAS fails to properly initiate pppd unless asterisk + is running as root + Reported by: Jaco Kroon + * [4f92dcd66b] Jaco Kroon -- dahdiras: Only set plugin dahdi.so to pppd + if we're running as root. + + Category: Applications/app_amd + + ASTERISK-28608: app_amd: Use time calculation to calculate timeout + Reported by: Michael Cargile + * [5bda460300] Michael Cargile -- app_amd: Fixed timeout issue + + Category: Applications/app_chanisavail + + ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to + deactivate CDR. + Reported by: Frederic LE FOLL + * [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail + sometimes fails to deactivate CDR. + ASTERISK-28527: ChanIsAvail() creates a CDR if unanswered=yes is set in + cdr.conf + Reported by: Frederic LE FOLL + * [2d0eee5418] Frederic LE FOLL -- ChanIsAvail() generates a CDR when + unanswered=yes in cdr.conf. + + Category: Applications/app_confbridge + + ASTERISK-28841: app_confbridge: Add support for disabling text messaging + for a user + Reported by: Joshua C. Colp + * [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling + text messaging. + ASTERISK-28790: Crash during conference call using confbridge and video + Reported by: Pascal Cadotte Michaud + * [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient + space for worst case NACK. + + Category: Applications/app_fax + + ASTERISK-28848: app_fax: Compile. + Reported by: Alexander Traud + * [26b8c99963] Alexander Traud -- app_fax: SpanDSP headers do not use + ast_malloc; ignore that. + + Category: Applications/app_meetme + + ASTERISK-28604: app_meetme, chan_ooh323 and cdr_mysql don't build on + 17.0.0 + Reported by: George Joseph + * [ed394ce5b1] Joshua C. Colp -- configure: Add check for MySQL client + bool and my_bool type usage. + * [a47cb71bb1] George Joseph -- Build: Fix compile issues with seldom + used modules + + Category: Applications/app_mixmonitor + + ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between + AMI MixMonitor and hangup + Reported by: Joshua C. Colp + * [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to + attach to hung up channels. + + Category: Applications/app_osplookup + + ASTERISK-28804: [patch] app_osplookup.c: Avoid a format truncation. + Reported by: Alexander Traud + * [527e4f6542] Alexander Traud -- app_osplookup: Avoid a format + truncation. + + Category: Applications/app_queue + + ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events + Reported by: Ove Aursand + * [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a + call as abandoned + ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned + Reported by: Kfir Itzhak + * [c83e4821e5] Kfir Itzhak -- app_queue: Fix leave-empty not recording a + call as abandoned + ASTERISK-29034: Lastpause of realtime members is reseting + Reported by: Evandro César Arruda + * [36dd15c659] Evandro César Arruda -- app_queue: Member lastpause time + reseting + ASTERISK-28951: Inconsistent behaviour queues.conf when there is (not) a + [general] section + Reported by: Walter Doekes + * [312c23b0e1] Walter Doekes -- app_queue: (Breaking change) + shared_lastcall and autofill default to no + ASTERISK-28950: Stale code in app_queue to check untouched channel + Reported by: Walter Doekes + * [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in + try_calling + ASTERISK-28644: Stale comment in app_queue about ring_entry exception + Reported by: Walter Doekes + * [db012e8cc6] Walter Doekes -- app_queue: Remove stale code in + try_calling + * [0e750cdd10] Walter Doekes -- app_queue: Fix old confusing comment + about when the members are called + ASTERISK-28952: Queue wrapuptime sometimes not respected (based on stale + lastcall time) + Reported by: Walter Doekes + * [0fb6738314] Walter Doekes -- app_queue: Read latest wrapuptime + instead of (possibly stale) copy + ASTERISK-28829: app_queue: leaking stasis subscription when Redirecting + call + Reported by: lvl + * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in + app_queue to avoid leaked stasis subscriptions + ASTERISK-25844: app_queue: Ghost channels in "core show channels" output + Reported by: Etienne Lessard + * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in + app_queue to avoid leaked stasis subscriptions + ASTERISK-28349: Pause reason not reported in QueueMember AMI event + Reported by: Niksa Baldun + * [9522390a69] Sean Bright -- app_queue: Deprecate the + QueueMemberPause.Reason field + + Category: Applications/app_record + + ASTERISK-28682: app_record: Lack of `beep` audio file causes application + to return error and hangup + Reported by: Corey Farrell + * [2f8b20b949] Corey Farrell -- app_record: Do not hang up if beep audio + is missing + + Category: Applications/app_system + + ASTERISK-28776: Non async-signal-safe syscalls used after fork before exec + Reported by: nappsoft + * [6b2d945174] Pirmin Walthert -- app.c: make sure that no + non-async-signal-safe syscalls are used after + + Category: Applications/app_voicemail + + ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it + is not sent by email/processed by the mailcmd command + Reported by: Leandro Dardini + * [b575868000] Sean Bright -- app_voicemail: Process urgent messages + with mailcmd + ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage + enabled and realtime voicemail_data is used + Reported by: Stas Kobzar + * [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving + message with realtime voicemail + ASTERISK-27622: empty voicemail.conf required for ARA (realtime) voicemail + to leave message + Reported by: Jim Van Meggelen + * [9be89d9913] Sean Bright -- app_voicemail: Set globals to default + values when voicemail.conf missing + + Category: Applications/app_voicemail/IMAP + + ASTERISK-28505: app_voicemail/IMAP: segfault in leave_voicemail because + not checking mailstream + Reported by: Alexei Gradinari + * [15624d9a7a] Alexei Gradinari -- app_voicemail/IMAP: check mailstream + not NULL in leave_voicemail + + Category: Applications/app_voicemail/ODBC + + ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage + enabled and realtime voicemail_data is used + Reported by: Stas Kobzar + * [ba8ccb9132] Sean Bright -- app_voicemail: Prevent crash when saving + message with realtime voicemail + + Category: Bridges/bridge_builtin_features + + ASTERISK-28920: bridge show all causes crash + Reported by: sungtae kim + * [25ae412f75] sungtae kim -- bridge.c: Fixed null pointer exception + + Category: Bridges/bridge_native_rtp + + ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility + check failure when negociated ptime is not default ptime. + Reported by: Frederic LE FOLL + * [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no + directmedia for ptime other than default ptime. + + Category: Bridges/bridge_softmix + + ASTERISK-28944: bridge_softmix: Transitioning a stream from inactive -> + sendrecv/sendonly doesn't re-negotiation + Reported by: Joshua C. Colp + * [8ad06394c4] Joshua C. Colp -- bridge_softmix: Add additional old + states for adding new source. + ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp + packets + Reported by: Jonathan Hunter + * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio + from mixed frame. + ASTERISK-28819: [patch] bridge_softmix_binaural: Show state in menuselect. + Reported by: Alexander Traud + * [7febd22304] Alexander Traud -- bridge_softmix_binaural: Show state in + menuselect. + ASTERISK-28618: bridge_softmix: hold not cleared when joining a softmix + bridge + Reported by: Kevin Harwell + * [e77cb32583] Kevin Harwell -- bridge_softmix: clear hold when joining + a softmix bridge + + Category: CDR/General + + ASTERISK-28677: CDR billsec is always 0 for transferred calls + Reported by: Maciej Michno + * [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when + leaving a parking bridge + ASTERISK-28636: app_chanisavail+cdr: ChanIsAvail sometimes fails to + deactivate CDR. + Reported by: Frederic LE FOLL + * [a83625b366] Frederic LE FOLL -- app_chanisavail/cdr: ChanIsAvail + sometimes fails to deactivate CDR. + ASTERISK-28566: CDR backend unload problem during active call(s) + Reported by: Marian Piater + * [51850a79ef] Sean Bright -- cdr_mysql: Don't clean up on unload unless + we can unregister from CDRs + + Category: CDR/cdr_pgsql + + ASTERISK-28571: cdr_pgsql: accesses obsolete (and finally removed) column + Reported by: Christoph Moench-Tegeder + * [52ade18420] Christoph Moench-Tegeder -- cdr_pgsql cel_pgsql + res_config_pgsql: compatibility with PostgreSQL 12 + + Category: Channels/chan_dahdi + + ASTERISK-28702: chan_dahdi: holding a channel via flash to dialtone times + out after 0:16:40 + Reported by: Andrew Siplas + * [5bd7281442] Andrew Siplas -- chan_dahdi: Change 999999 to INT_MAX to + better reflect "no timeout" + ASTERISK-28615: chan_dahdi: PRI span status may stay "Down, Active" after + a short alarm + Reported by: Frederic LE FOLL + * [a68299f508] Frederic LE FOLL -- chan_dahdi: PRI span status may stay + "Down, Active" after a short alarm + ASTERISK-28536: Asterisk release candidates fail to build on FreeBSD + Reported by: Guido Falsi + * [4072e219f7] Guido Falsi -- chan_dahdi: Fix build with clang/llvm + ASTERISK-28525: chan_dahdi: set CHANNEL(hangupsource) when a PRI channel + hangs up + Reported by: Frederic LE FOLL + * [41b67f150e] Frederic LE FOLL -- chan_dahdi: set CHANNEL(hangupsource) + when a PRI channel hangs up + + Category: Channels/chan_local + + ASTERISK-28938: core_unreal / core_local: Add support for multistream and + re-negotiation + Reported by: Joshua C. Colp + * [de2813cf23] Joshua C. Colp -- core_unreal / core_local: Add + multistream and re-negotiation. + ASTERISK-25844: app_queue: Ghost channels in "core show channels" output + Reported by: Etienne Lessard + * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in + app_queue to avoid leaked stasis subscriptions + + Category: Channels/chan_pjsip + + ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 + Reported by: Joseph Ades + * [31fbfc5e95] Kevin Harwell -- chan_pjsip: disallow + PJSIP_SEND_SESSION_REFRESH pre-answer execution + * [4eba6b9eb2] Kevin Harwell -- PJSIP_MEDIA_OFFER: override + configuration on refresh + ASTERISK-28886: chan_pjsip: PJSIP_SC_NULL does not exist in pjproject + 2.7.2 + Reported by: Jared Smith + * [8b925fbda3] Kevin Harwell -- chan_pjsip: don't use PJSIP_SC_NULL as + it only exists pjproject 2.8+ + ASTERISK-28923: T.38 Segfaults in chan_pjsip_queryoption + Reported by: Yury Kirsanov + * [41f3a7da4d] George Joseph -- res_fax: Don't start a gateway if either + channel is hung up + ASTERISK-28835: IPv6 addresses in SDP incorrectly formatted + Reported by: Daniel Heckl + * [9f117ac9ef] Daniel Heckl -- res_pjsip: Fixed format of IPv6 addresses + for external media addresses + ASTERISK-28817: chan_pjsip: constant DTMF tone if RTP is not setup yet + Reported by: Kevin Harwell + * [fa3c8f94e0] Kevin Harwell -- chan_pjsip: digit_begin - constant DTMF + tone if RTP is not setup yet + ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during + direct-media (native_rtp) bridge + Reported by: Michael Neuhauser + * [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore + rtptimeout if direct-media is active + ASTERISK-28759: A non negotiated rtp frame causes call disconnection when + there is a SSRC change + Reported by: Paulo Vicentini + * [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when + remote SSRC changes. + ASTERISK-28766: PJSIP blind transfer not completed after using + Proceeding() + Reported by: lvl + * [d1a2ff0aaf] lvl -- res_pjsip_refer: ensure refer progress is still + sent after Proceeding() + ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" + field + Reported by: Jean Aunis - Prescom + * [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text + message variables work in ARI + ASTERISK-28492: pjsip reload not reloading wizard endpoint/pickup_group + endpoint/call_group + Reported by: Jean-Denis Girard + * [b40dd11afe] Sean Bright -- res_pjsip_config_wizard: Fix change + detection for wizard settings + ASTERISK-28502: chan_pjsip incorrectly re-writes REGISTER 200 Response + Contact + Reported by: Ross Beer + * [cbc1136704] George Joseph -- res_pjsip_nat: Restore original contact + for REGISTER responses + ASTERISK-28578: race condition on pjsip channelstats command + Reported by: Salah Ahmed + * [ddb0091da5] Salah Ahmed -- Crash during "pjsip show channelstats" + execution + ASTERISK-28561: Asterisk Deadlocks + Reported by: Aheliotech + * [bf6f27388d] Joshua Colp -- pbx: deadlock when outgoing dialed channel + hangs up too quickly + ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI + Reported by: Jeremiah Gadd + * [c03f50c1c8] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF + on hungup channel + ASTERISK-28538: chan_pjsip: Deadlock on fax detection + Reported by: Joshua C. Colp + * [c358da472e] Joshua Colp -- chan_pjsip: Relock correct channel during + "fax" redirect. + + Category: Channels/chan_sip/General + + ASTERISK-29011: chan_sip: ToHost property not cleared on reload + Reported by: Dennis + * [9058d9e591] Dennis Buteyn -- chan_sip: Clear ToHost property on peer + when changing to dynamic host + ASTERISK-28957: chan_sip: chan_sip does not process 400 response to an + INVITE. + Reported by: Frederic LE FOLL + * [a423f935c9] Frederic LE FOLL -- chan_sip: chan_sip does not process + 400 response to an INVITE. + ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp + packets + Reported by: Jonathan Hunter + * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio + from mixed frame. + ASTERISK-28651: chan_sip logs errors on tx to non-existent TCP connections + Reported by: Jaco Kroon + * [365d007eb6] Jaco Kroon -- chan_sip: in case of tcp/tls, be less + annoying about tx errors. + ASTERISK-28647: chan_sip: RTP frames not transmitted after emitting a COLP + Reported by: Jean Aunis - Prescom + * [9c9296c635] Jean Aunis -- chan_sip: voice frames are no longer + transmitted after emitting a COLP + ASTERISK-28637: chan_sip+native_bridge_rtp: directmedia compatibility + check failure when negociated ptime is not default ptime. + Reported by: Frederic LE FOLL + * [7624cbb155] Frederic LE FOLL -- chan_sip+native_bridge_rtp: no + directmedia for ptime other than default ptime. + + Category: Channels/chan_sip/Interoperability + + ASTERISK-28718: chan_sip: Returns 403 if RTP ports are depleted, should + return 503 + Reported by: Walter Doekes + * [43620cbf6c] Walter Doekes -- chan_sip: Return 503 if we're out of RTP + ports + ASTERISK-28686: chan_sip strictrtp=yes fails when media source is changed: + no audio + Reported by: Walter Doekes + * [711a3fed56] Walter Doekes -- chan_sip: Always process updated SDP on + media source change + + Category: Channels/chan_sip/Messaging + + ASTERISK-28693: chan_sip: SIP MESSAGE beginning with a whitespace appears + empty in the dialplan + Reported by: Frank Matano + * [f309b86e36] Sean Bright -- chan_sip.c: Stop handling continuation + lines after reading headers + + Category: Channels/chan_sip/TCP-TLS + + ASTERISK-28372: Asterisk REPLY Wrong Contact header port (TCP) + Reported by: Anton Satskiy + * [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with + non-default TCP/TLS ports. + ASTERISK-24428: Document that Asterisk will use the default SIP ports + (5060 for TCP, 5061 for TLS) if the extern option variants aren't used + Reported by: sstream + * [52f07176b6] Alexander Traud -- chan_sip: externhost/externaddr with + non-default TCP/TLS ports. + ASTERISK-27195: chan_sip: only sets ToS bits on UDP socket, ignoring TCP + and TLS sockets + Reported by: Joshua Roys + * [4d0ab620be] Alexander Traud -- chan_sip: DiffServ/ToS not only on UDP + but also on TCP and TLS sockets. + + Category: Channels/chan_sip/Transfers + + ASTERISK-28677: CDR billsec is always 0 for transferred calls + Reported by: Maciej Michno + * [6818c3d1d2] George Joseph -- cdr.c: Set event time on party b when + leaving a parking bridge + + Category: Channels/chan_unistim + + ASTERISK-28803: [patch] chan_unistim: Avoid tautological warnings with + clang. + Reported by: Alexander Traud + * [b38f664250] Alexander Traud -- chan_unistim: Avoid tautological + warnings with clang. + ASTERISK-25592: chan_unistim: Clang Warning: variable sized type not at + end of a struct + Reported by: Alexander Traud + * [3863ab9af9] Igor Goncharovsky -- chan_unistim: Fix clang warning: + variable sized type not at end of a struct + + Category: Codecs/codec_resample + + ASTERISK-28511: codec_resample: Bad sound quality when up sampling from + SLIN16 to SLIN32 + Reported by: Ruddy G + * [e4289b9e56] Sean Bright -- codec_resample: Ensure OUTSIDE_SPEEX is + defined when necessary + * [b096389660] Sean Bright -- codec_resample: Upgrade speex_resample to + fix up-sampling bug + + Category: Codecs/codec_silk + + ASTERISK-28706: silk 24hHz doesn't show up in 'core show translation' + output + Reported by: Sean Bright + * [dfad69ce7c] Sean Bright -- translate.c: Fix silk 24kHz truncation in + 'core show translation' + + Category: Configs/Basic-PBX + + ASTERISK-28667: Asterisk ignores parsing of config files if a Byte order + mark is present + Reported by: Robin Leffmann + * [40b5cf8f52] Sean Bright -- config.c: Skip UTF-8 BOMs if present when + reading config files + + Category: Configs/Samples + + ASTERISK-29123: logger.conf.sample missing comment mark on line 115 + Reported by: Andrew Siplas + * [79d749d2b5] Andrew Siplas -- logger.conf.sample: add missing comment + mark + + Category: Contrib/General + + ASTERISK-27243: contrib: valgrind.supp doesn't suppress what it's supposed + to due to invalid syntax + Reported by: Richard Kenner + * [095c204fe0] snuffy -- contrib/valgrind: Fix use of frame-level + suppression + ASTERISK-28664: "trustrpid" is misspelled in sip_to_pjsip.py + Reported by: Pascal Cadotte Michaud + * [e494d5fd76] Pascal Cadotte Michaud -- sip_to_pjsip.py: Fix trustrpid + typo + + Category: Core/ACL + + ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault + on reading rule from realtime + Reported by: Andrew Yager + * [7a43bedd72] Sean Bright -- acl.c: Coerce a NULL pointer into the + empty string + + Category: Core/Bridging + + ASTERISK-28841: app_confbridge: Add support for disabling text messaging + for a user + Reported by: Joshua C. Colp + * [6cfc6ff53c] Joshua C. Colp -- confbridge: Add support for disabling + text messaging. + + Category: Core/BuildSystem + + ASTERISK-28929: pjproject_bundled: Honor --without-pjproject. + Reported by: Alexander Traud + * [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor + --without-pjproject. + ASTERISK-28837: pjproject_bundled: Honor --without-pjproject. + Reported by: Alexander Traud + * [966acc6251] Alexander Traud -- pjproject_bundled: Honor + --without-pjproject. + ASTERISK-28824: BuildSystem: Search for Python/C API when possibly needed + only. + Reported by: Alexander Traud + * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API + when possibly needed only. + ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming + Language is python-2.7. + Reported by: Alexander Traud + * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API + when possibly needed only. + ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. + Reported by: Alexander Traud + * [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and + doc/pdf leftovers. + ASTERISK-28818: [patch] BuildSystem: Allow space in path. + Reported by: Alexander Traud + * [7a04947abd] Alexander Traud -- BuildSystem: Allow space in path. + ASTERISK-28487: compile menuselect on gentoo + Reported by: Kilburn + * [e40f248fac] Sean Bright -- menuselect: Fix curses build on Gentoo + Linux + + Category: Core/Channels + + ASTERISK-25844: app_queue: Ghost channels in "core show channels" output + Reported by: Etienne Lessard + * [f217fcdc62] Nathan Bruning -- app_queue: track masquerades in + app_queue to avoid leaked stasis subscriptions + ASTERISK-28795: channel: write to a stream on multi-frame writes + Reported by: Kevin Harwell + * [3c345ec56d] Kevin Harwell -- channel: write to a stream on + multi-frame writes + ASTERISK-28499: translate: Crash when frame does not have a "src" field + set + Reported by: Gregory Massel + * [1e9714a050] Joshua Colp -- AST-2019-005 - translate: Don't assume all + frames will have a src. + + Category: Core/Configuration + + ASTERISK-28955: "setvar" doesn't work properly in dahdi-channels.conf + Reported by: Marin Odrljin + * [d88e230037] Guido Falsi -- chan_dadhi: Fix setvar in dahdi channels + ASTERISK-23756: setvar directive when used in template and a child of said + template, results in duplicate variable names + Reported by: Michael Goryainov + * [32ce6e9a06] Michael Goryainov -- channels: Allow updating variable + value + + Category: Core/General + + ASTERISK-28797: [patch] tcptls: Fix notice when TLS is enabled but not + configured. + Reported by: Alexander Traud + * [f9ea75d117] Alexander Traud -- tcptls: Fix notice when TLS is enabled + but not supported. + ASTERISK-28839: Sporadic crashes with Segmentation fault + Reported by: Joeran Vinzens + * [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP + re-negotiation scenarios. + ASTERISK-28780: app_mixmonitor: Memory leak due to race condition between + AMI MixMonitor and hangup + Reported by: Joshua C. Colp + * [98d10d0a16] Joshua C. Colp -- audiohook: Don't allow audiohooks to + attach to hung up channels. + ASTERISK-28498: cel / cdr: Event times may be incorrect + Reported by: Joshua C. Colp + * [261646c1c4] Joshua Colp -- cdr / cel: Use event time at event + creation instead of processing. + + Category: Core/RTP + + ASTERISK-28480: json integer overflow in ssrc and timestamp + Reported by: Salah Ahmed + * [3656c42cb0] Kevin Harwell -- various modules: json integer overflow + + Category: Core/Stasis + + ASTERISK-28755: SIP/Stasis: SIP headers not transmitted in the "variables" + field + Reported by: Jean Aunis - Prescom + * [a715cf5aaa] Kevin Harwell -- message & stasis/messaging: make text + message variables work in ARI + ASTERISK-28553: stasis.c: Crash during unload + Reported by: Kevin Harwell + * [729b286d59] Joshua Colp -- stasis: Pass bumped topic_all reference to + proxy_dtor. + + Category: Core/Streams + + ASTERISK-28870: streams: One memory leak and one issue cloning streams + Reported by: George Joseph + * [7fbfbe7da0] George Joseph -- streams: Fix one memory leak and one + formats ref issue + ASTERISK-28846: stream: Enforce formats immutability + Reported by: Joshua C. Colp + * [1c5e68580a] Joshua C. Colp -- stream: Enforce formats immutability + and ensure formats exist. + ASTERISK-28625: Playback of local files impacted by large media cache + Reported by: Kevin Reeves + * [c626ccec12] Kevin Reeves -- main/file.c: Limit media cache usage to + remote files. + + Category: Core/UDPTL + + ASTERISK-28483: packet lost on UDPTL wrap around + Reported by: Torrey Searle + * [084901d548] Torrey Searle -- main/udptl.c: correctly handle udptl + sequence wrap around + + Category: Documentation + + ASTERISK-28816: [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers. + Reported by: Alexander Traud + * [7cdb493a1e] Alexander Traud -- BuildSystem: Remove doc/tex and + doc/pdf leftovers. + ASTERISK-24484: Update documentation for statsd module - usage + requirements unclear + Reported by: Dan Jenkins + * [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does + nothing on its own + ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for + hostnames + Reported by: Joshua C. Colp + * [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document + support for hostnames + ASTERISK-28507: Wiki docs missing for MessageWaiting + Reported by: David M. Lee + * [d5f3ec92d0] George Joseph -- CI: Update buildAsterisk.sh to do a + "make full" + + Category: Functions/General + + ASTERISK-28626: Missing arguments in PJSIP_CONTACT function documentation + Reported by: Pascal Cadotte Michaud + * [bf4dd3d837] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing + argument documentation + * [7e3015d779] Pascal Cadotte Michaud -- PJSIP_CONTACT: add missing + argument documentation + ASTERISK-26481: FILE function grabs garbage along with read data when + target line has no newline + Reported by: Jonathan Harris + * [bf7c808604] Sean Bright -- func_env: Prevent FILE() from reading + garbage at end-of-file + + Category: Functions/func_aes + + ASTERISK-28788: func_aes: incorrectly printing error 'declined to load' + Reported by: Alexander Traud + * [cd8cbf7384] Alexander Traud -- func_aes: Avoid incorrect error + message on load. + + Category: Functions/func_channel + + ASTERISK-28796: func_channel: cannot read fields exten, context, + userfield, channame from dialplan + Reported by: Sébastien Duthil + * [d40e343710] Sebastien Duthil -- func_channel: allow reading 4 fields + from dialplan + + Category: Functions/func_enum + + ASTERISK-26711: func_enum: ENUM code wrong case + Reported by: Vitold + * [517224ce85] Sean Bright -- enum.c: Add support for regular expression + flag in NAPTR record + ASTERISK-19460: [patch] Function TXTCIDNAME never actually makes DNS calls + and always returns an empty string + Reported by: George Joseph + * [ab63f0cd0f] Sean Bright -- enum.c: Make ast_get_txt() actually do + something. + + Category: Functions/func_odbc + + ASTERISK-20325: Comments in configs/func_odbc.conf.sample are not + consistent with examples. Missing examples. + Reported by: Olivier Krief + * [c4e0983742] Sean Bright -- func_odbc.conf.sample: Clarify sample + documentation + ASTERISK-28497: func_odbc: truncating Unicode string on readsql + Reported by: Boris P. Korzun + * [8979921da9] Boris P. Korzun -- func_odbc: acf_odbc_read() and + cli_odbc_read() unicode support + + Category: Functions/func_version + + ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified + versions + Reported by: cmaj + * [543f936147] cmaj -- Makefile: Fix certified version numbers + + Category: General + + ASTERISK-28930: ./configure --without-ssl build failure + Reported by: Jaco Kroon + * [9b5042433b] Joshua C. Colp -- menuselect: Resolve infinite loop in + dependency scenario. + ASTERISK-28838: AST_MODULE_INFO requires, MODULEINFO does not mention + Reported by: Alexander Traud + * [abf4d74384] Alexander Traud -- cdr_odbc: Sync load- and build-time + deps. + * [191f136260] Alexander Traud -- res_pjsip_refer: Add build-time + dependency. + * [5c2b8fdeca] Alexander Traud -- app_getcpeid: Add build-time + dependency. + * [008f46bf1e] Alexander Traud -- res_pjsip: Sync load- and build-time + deps. + * [e2affa3b0a] Alexander Traud -- curl: Add build-time dependency. + * [f1135b453b] Alexander Traud -- res_pjsip: Add build-time dependency. + ASTERISK-28609: Memory Leak in res_rtp_asterisk.c + Reported by: Ted G + * [39c920ac78] George Joseph -- res_rtp_asterisk: Add frame list + cleanups to ast_rtp_read + ASTERISK-28590: utils.c throws repeated warnings; + "pthread_attr_setstacksize: Invalid argument" + Reported by: Speed Dial Dave + * [a4222614c4] Sean Bright -- utils.h: Set lower bound for thread stack + size to PTHREAD_STACK_MIN + ASTERISK-28523: Asterisk 16.5.0 Memory leak + Reported by: Cyril Ramière + * [a4caaef64c] Kevin Harwell -- res_sorcery_memory_cache: stale item + update leak + ASTERISK-28472: Asterisk occasionally passes a NULL as srtp->session to + srtp_protect/unprotect causing SEGV + Reported by: Jonas Swiatek + * [b805e1237d] Kevin Harwell -- srtp: Fix possible race condition, and + add NULL checks + + Category: PBX/General + + ASTERISK-29046: pbx: Deadlock when doing a reload, while simultaneously + doing an ExtensionState on a pattern match hint that ends up adding an + extension + Reported by: Ramarajan + * [6d50d152d8] Joshua C. Colp -- pbx: Fix hints deadlock between reload + and ExtensionState. + ASTERISK-28695: core: minmemfree watermark uses free RAM, not available + RAM + Reported by: Kevin Flyn + * [50d02d6194] Sean Bright -- pbx.c: Include filesystem cache in free + memory calculation + ASTERISK-28605: chan_dahdi: Deadlock in Hangup Scenarios with concurrent + command pri show span X + Reported by: Dirk Wendland + * [ee7d72eb72] George Joseph -- sig_pri: Fix deadlock caused by + sig_pri_queue_hangup + + Category: PBX/pbx_config + + ASTERISK-28534: Segmentation fault when there is no priority for an + extension + Reported by: Timothy Vanderaerden + * [702019fc80] Sean Bright -- pbx: Prevent Realtime switch crash on + invalid priority + + Category: PBX/pbx_dundi + + ASTERISK-21205: [patch] dundi_read_result crash due to negative number + Reported by: Jaco Kroon + * [40e93b0240] Jaco Kroon -- dundi: fix NULL dereference. + + Category: Resources/res_ari + + ASTERISK-28948: ARI channel create doesn't referencing the channel_id + parameter + Reported by: sungtae kim + * [bbe0f2230d] sungtae kim -- res_ari: Fix create channel request + channelId parameter parsing + ASTERISK-28679: stasis application is destroyed after its creation + Reported by: Francois Blackburn + * [4206830a52] Kevin Harwell -- res_stasis: trigger cleanup after update + ASTERISK-28585: ari/resource_events: Crash in event session cleanup + Reported by: Kevin Harwell + * [360936ead5] Joshua Colp -- res_ari_events: Add module reference when + a WebSocket is open. + + Category: Resources/res_ari_bridges + + ASTERISK-28898: bridge_softmix: Conference bridge not passing silent rtp + packets + Reported by: Jonathan Hunter + * [e8c8d69d47] Joshua C. Colp -- bridge_softmix: Always remove audio + from mixed frame. + + Category: Resources/res_ari_channels + + ASTERISK-28940: /channels/create doesn't get any parameters from the body + Reported by: sungtae kim + * [fa7c69f40f] sungtae kim -- res_ari: Fix create request body parameter + parsing. + ASTERISK-28847: ARI channels cuts the endpoint string over 80 characters + Reported by: sungtae kim + * [9ad3d2829c] sungtae kim -- res_ari_channels: Fixed endpoint 80 + characters limit + + Category: Resources/res_calendar_exchange + + ASTERISK-28572: Memory leaks in res_calendar_exchange and + res_calendar_icalendar + Reported by: Yoooooo Ha + * [16e668c7dd] Sean Bright -- res_calendar: Resolve memory leak on + calendar destruction + + Category: Resources/res_calendar_icalendar + + ASTERISK-28572: Memory leaks in res_calendar_exchange and + res_calendar_icalendar + Reported by: Yoooooo Ha + * [16e668c7dd] Sean Bright -- res_calendar: Resolve memory leak on + calendar destruction + + Category: Resources/res_corosync + + ASTERISK-28888: res_corosync: causes asterisk crash in huge distributed + environment. + Reported by: Università di Bologna - CESIA VoIP + * [0c1c386634] Università di Bologna - CESIA VoIP -- res_corosync: Fix + crash in huge distributed environment. + + Category: Resources/res_fax + + ASTERISK-28900: res_fax: Double frame free when gateway in use with + off-nominal format usage + Reported by: Gregory Massel + * [d2500c6273] Joshua C. Colp -- res_fax: Don't consume frames given to + fax gateway on write. + ASTERISK-28660: res_fax: wrap Asterisk initiated negotiation with config + option + Reported by: Kevin Harwell + * [b6f5607359] Kevin Harwell -- res_fax: wrap v21 detected Asterisk + initiated negotiation with config option + + Category: Resources/res_http_websocket + + ASTERISK-28975: res_http_websocket: Text payload data doesn't necessary + include trailing zero + Reported by: Nickolay V. Shmyrev + * [e4d24f5137] Nickolay Shmyrev -- res_http_websocket: Avoid reading + past end of string + ASTERISK-28562: SIP WSS message not processed until next frame arrives + Reported by: Robert Sutton + * [87110c1bdf] Sean Bright -- websocket: Consider pending SSL data when + waiting for socket input + + Category: Resources/res_musiconhold + + ASTERISK-28927: Asterisk crash in music on hold + Reported by: David Cunningham + * [57554c2834] Sean Bright -- res_musiconhold.c: Prevent crash with + realtime MoH + ASTERISK-28892: res_musiconhold: Module res_musiconhold throws false + warning + Reported by: Nicholas John Koch + * [fef97a9a72] Nicholas John Koch -- res_musiconhold: Added check for + dot character in path of playlist entries to avoid warnings + ASTERISK-28735: Realtime MoH Unknown format '' -- defaulting to SLIN + Reported by: Ross Beer + * [aeff1f2c53] Sean Bright -- res_musiconhold: Avoid spurious warning + when 'format' is the empty string + + Category: Resources/res_parking + + ASTERISK-29042: res_parking: Parker UUID is no longer copied + Reported by: Misha Vodsedalek + * [4f0766dcda] Joshua C. Colp -- parking: Copy parker UUID as well. + ASTERISK-28631: res_parking: Doesn't park when parkee and parker are the + same + Reported by: Ross Beer + * [811ae88da4] Joshua Colp -- parking: Fall back to parker channel name + even if it matches parkee. + ASTERISK-28616: parking: Deadlock when multi call parking + Reported by: Joshua C. Colp + * [807a70b7ae] Joshua Colp -- parking: Fix case where we can't get the + parker. + * [e924c5107c] Joshua Colp -- parking: Use channel snapshot instead of + channel. + + Category: Resources/res_pjsip + + ASTERISK-28995: res_pjsip_registrar: Expires on statically configured + contacts is not correct + Reported by: tootai + * [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an + expiration for static contacts. + ASTERISK-28965: res_pjsip: Apply outbound proxy to static contacts on AOR + Reported by: Joshua C. Colp + * [4f86118bd8] Joshua C. Colp -- res_pjsip: Apply AOR outbound proxy to + static contacts. + ASTERISK-28936: res_pjsip: crash when dialing non-sip uri + Reported by: Walter Doekes + * [e74dde5100] Walter Doekes -- pjsip: Prevent invalid memory access + when attempting to contact a non-sip URI + ASTERISK-28794: res_pjsip: Crash when escaping during URI printing + Reported by: nappsoft + * [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing + the contact_user value. + ASTERISK-26780: res_pjsip: PJSIP Registration Fails when + transport=transport-udp6 + Reported by: Peter Sokolov + * [c8dec423d2] Peter Sokolov -- pjsip_resolver.c: Ensure AAAA dns + requests are made. + ASTERISK-28854: SIGSEGV when pjsip show history encounters IPV6 address + Reported by: Roger James + * [4a072c4890] Roger James -- res_pjsip_history.c: Fix to stop SIGSEGV + when IPv6 addresses are encountered. + ASTERISK-28056: res_pjsip: Incorrect endpoint status after endpoint + synchronization for a specific AOR + Reported by: Jason Hord + * [d845464c76] Jason Hord -- res_pjsip: Don't set endpoint to + unavailable in all cases. + ASTERISK-28790: Crash during conference call using confbridge and video + Reported by: Pascal Cadotte Michaud + * [96e8d411e1] Joshua C. Colp -- res_rtp_asterisk: Ensure sufficient + space for worst case NACK. + ASTERISK-28743: Asterisk is crashing if the 200 OK with SDP + Reported by: sungtae kim + * [8147f43756] Sungtae Kim -- res_pjsip_session: Fixed wrong session + termination + ASTERISK-23407: Fix the FSF address in the headers of lots of pjproject + files + Reported by: Jared Smith + * [0a7fe3097f] Jared Smith -- indications.conf.sample: Add indication + tones for Indonesia + ASTERISK-28139: RTP Stream Incorrect Payload Type Causes Asterisk To Drop + Calls + Reported by: Paul Brooks + * [313189aae2] Sean Bright -- chan_pjsip: Ignore RTP that we haven't + negotiated + ASTERISK-28641: res_pjsip Segfaults when realtime configuration to an AOR + points to a not existent AOR + Reported by: Ross Beer + * [b1be06df8d] Sean Bright -- res_pjsip_registrar.c: Prevent potential + double free if AOR is not found + ASTERISK-28544: Wrong contact representation in ipv6 mode + Reported by: Jørgen H + * [377d7bdab6] Sean Bright -- res_pjsip_transport_websocket: Don't put + brackets around local_name if IPv6 + ASTERISK-28521: pjsip: Memory Leak + Reported by: Mark + * [cc83e76aa5] George Joseph -- pjproject_bundled: Revert pjproject 2.9 + commits causing leaks + ASTERISK-28228: res_pjsip: pjsip show contacts prints double entries + Reported by: Ian Jones + * [86452c9fa4] Joshua Colp -- res_pjsip: Fix multiple of the same + contact in "pjsip show contacts". + + Category: Resources/res_pjsip_acl + + ASTERISK-28697: res_pjsip: Named ACL does not update on reload if changed + Reported by: Timothy Vanderaerden + * [d6712790cd] Joshua C. Colp -- pjsip: Update ACLs on named ACL + changes. + + Category: Resources/res_pjsip_diversion + + ASTERISK-29001: chan_pjsip does not process or forward 181 responses + Reported by: Torrey Searle + * [addd295cda] Torrey Searle -- res_pjsip_diversion: handle 181 + + Category: Resources/res_pjsip_endpoint_identifier_ip + + ASTERISK-25429: res_pjsip_endpoint_identifier_ip: Document support for + hostnames + Reported by: Joshua C. Colp + * [29d867ed67] Sean Bright -- res_pjsip_endpoint_identifier_ip: Document + support for hostnames + + Category: Resources/res_pjsip_logger + + ASTERISK-28932: res_pjsip_logger writing too big packets + Reported by: nappsoft + * [e8c6e9ae5d] Pirmin Walthert -- res_pjsip_logger: use the correct + pointer when logging tx_messages to pcap + ASTERISK-28921: Wrong return value check for fwrite when writing to pcap + file + Reported by: nappsoft + * [c16937cdbe] Pirmin Walthert -- res_pjsip_logger.c: correct the return + value checks when writing to pcap + + Category: Resources/res_pjsip_messaging + + ASTERISK-26082: res_pjsip_messaging: MessageSend Content-Type can't be + changed + Reported by: Alex + * [03d24ca4c1] Sean Bright -- res_pjsip_messaging: Allow Content-Type to + be overridden + ASTERISK-25421: PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the + error when sending + Reported by: Dmitriy Serov + * [b1ca2c5d71] Sean Bright -- res_pjsip_messaging: Ensure + MESSAGE_SEND_STATUS is set properly + + Category: Resources/res_pjsip_mwi + + ASTERISK-28575: MWI Send Notify Crash on 16.6 + Reported by: Joshua Elson + * [5dae803eea] Kevin Harwell -- res_pjsip_mwi: potential double unref, + and potential unwanted double link + ASTERISK-28552: res_pjsip_mwi: Frack during unload on unsolicited_mwi + container + Reported by: Kevin Harwell + * [12dbeb69b0] Kevin Harwell -- res_pjsip_mwi: use an ao2_global object + for mwi containers + + Category: Resources/res_pjsip_nat + + ASTERISK-28884: x-ast-orig-host not filtered out from request URI and To + header + Reported by: nappsoft + * [1399f8b4fe] Pirmin Walthert -- res_pjsip_nat.c: remove + x-ast-orig-host from request URI and To header + + Category: Resources/res_pjsip_notify + + ASTERISK-27775: res_pjsip_notify: Multiple Event headers can be present + instead of just one + Reported by: AvayaXAsterisk + * [90af050fa4] Sean Bright -- res_pjsip_notify: Only allow a single + Event header to be added to a NOTIFY + + Category: Resources/res_pjsip_outbound_registration + + ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first + entry in a SRV record set + Reported by: George Joseph + * [78b01f41ae] George Joseph -- res_pjsip_outbound_registration: Fix SRV + failover on timeout + ASTERISK-28624: res_pjsip_outbound_registration: add SRV failover + Reported by: Kevin Harwell + * [d5d41409e2] Kevin Harwell -- res_pjsip_outbound_registration: add + support for SRV failover + ASTERISK-28521: pjsip: Memory Leak + Reported by: Mark + * [cc83e76aa5] George Joseph -- pjproject_bundled: Revert pjproject 2.9 + commits causing leaks + + Category: Resources/res_pjsip_path + + ASTERISK-28463: res_pjsip_path: Crash when invalid contact is configured + Reported by: Juan Martin + * [982a5025b3] Sean Bright -- res_pjsip_registrar: Validate Contact URI + before adding to responses + + Category: Resources/res_pjsip_pubsub + + ASTERISK-28714: REGRESSION: Feature subscription_persistence_recreate + (ASTERISK-27759) Causes Segfaults + Reported by: Ross Beer + * [a1f0c833ab] Joshua C. Colp -- res_pjsip_pubsub: Increment persistence + data ref when recreating. + ASTERISK-27759: res_pjsip_pubsub: Subscription persistence does not + preserve XML version number + Reported by: Bryan Nelson + * [4e7adbd8f4] Joshua C. Colp -- res_pjsip_pubsub: Add ability to + persist generator state information. + + Category: Resources/res_pjsip_registrar + + ASTERISK-28995: res_pjsip_registrar: Expires on statically configured + contacts is not correct + Reported by: tootai + * [99eafe5771] Joshua C. Colp -- res_pjsip_registrar: Don't specify an + expiration for static contacts. + + Category: Resources/res_pjsip_sdp_rtp + + ASTERISK-28784: res_pjsip_sdp_rtp: Only do hold/unhold on first audio + stream + Reported by: Joshua C. Colp + * [34750d2068] Joshua C. Colp -- res_pjsip_sdp_rtp: Only do hold/unhold + on default audio stream. + ASTERISK-28774: chan_pjsip's rtptimeout is erroneously triggered during + direct-media (native_rtp) bridge + Reported by: Michael Neuhauser + * [5562fb2ea0] Michael Neuhauser -- chan_psip, res_pjsip_sdp_rtp: ignore + rtptimeout if direct-media is active + ASTERISK-28754: ASTERISK-28738 Causes Audio Issue After Hold + Reported by: Ross Beer + * [77c9ba8e63] Torrey Searle -- res/res_pjsip_sdp_rtp: Fix MOH + transitions + ASTERISK-28738: Incorrect state machine used when MOH_PASSTHRU is used + Reported by: Torrey Searle + * [bf4340f0ec] Torrey Searle -- res_pjsip_sdp_rtp: implement hold state + handling on moh_passthrough + ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media + stream if codecs create additional streams and offer does not have them + Reported by: nappsoft + * [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on + created streams for incoming SDP. + + Category: Resources/res_pjsip_session + + ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due + to codec negotiation after upgrading from Asterisk 16 + Reported by: Ross Beer + * [5cc4a391b3] Joshua C. Colp -- res_pjsip: Adjust outgoing offer call + pref. + ASTERISK-29033: res_pjsip_session: Aggressively terminates session on + failed re-INVITE + Reported by: Joshua C. Colp + * [3c074038fe] Joshua C. Colp -- res_pjsip_session: Don't aggressively + terminate on failed re-INVITE. + ASTERISK-28953: res_pjsip_session: Preserve stream label + Reported by: Joshua C. Colp + * [ee8ea9275f] Joshua C. Colp -- res_pjsip_session: Preserve label on + incoming re-INVITE. + ASTERISK-28871: res_pjsip_session: Unnecessary re-Invite on call answer + Reported by: Alexei Gradinari + * [afa2c9a868] Joshua C. Colp -- bridge: Don't try to match audio + formats. + ASTERISK-28783: res_pjsip_session: Allow default non-audio streams to have + reflected state + Reported by: Joshua C. Colp + * [9620ecbf80] Joshua C. Colp -- res_pjsip_session: Don't restrict + non-audio default streams to sendrecv. + ASTERISK-28730: res_pjsip_session: Fix out of order session refreshes + Reported by: Joshua C. Colp + * [ac155decae] Joshua C. Colp -- res_pjsip_session: Fix off-nominal + session refreshes. + ASTERISK-28659: res_pjsip_sdp_rtp: Bundle includes non-existent media + stream if codecs create additional streams and offer does not have them + Reported by: nappsoft + * [a603d7d324] Joshua C. Colp -- res_pjsip_session: Set stream state on + created streams for incoming SDP. + ASTERISK-28445: res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on + hangup when TEST_FRAMEWORK enabled + Reported by: Bernhard Schmidt + * [6ee1f1f507] Sean Bright -- res_pjsip_session.c: Prevent + use-after-free with TEST_FRAMEWORK enabled + ASTERISK-28086: chan_pjsip: Crash when initiating PlayDTMF over AMI + Reported by: Jeremiah Gadd + * [c03f50c1c8] lvl -- chan_pjsip: Prevent segfault when running PlayDTMF + on hungup channel + + Category: Resources/res_pjsip_t38 + + ASTERISK-28621: Enforce T.38 error correction mode at 200 ok received + Reported by: Salah Ahmed + * [330ffa2bce] Salah Ahmed -- res_pjsip_t38: T.38 error correction mode + selection at 200 ok received + + Category: Resources/res_realtime + + ASTERISK-21794: CLI command 'realtime update2' syntax failure when using + according to usage help + Reported by: Cedric BASSAGET + * [094e87b0dc] Sean Bright -- res_realtime: Fix 'realtime update2' + argument handling + + Category: Resources/res_rtp_asterisk + + ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string + to each message block. + Reported by: Thomas Johnson + * [5ec7099312] Sean Bright -- bridge_channel: Ensure text messages are + zero terminated + ASTERISK-28939: res_rtp_asterisk: Don't have send/receive buffers on + non-WebRTC + Reported by: Joshua C. Colp + * [c84d962eae] Joshua C. Colp -- res_rtp_asterisk: Don't assume setting + retrans props means to enable. + ASTERISK-28904: RTP ICE leaks the memory + Reported by: sungtae kim + * [c8c94b6cf1] sungtae kim -- res_rtp_asterisk.c: Fixed memory leak + ASTERISK-28852: Unprotected access to nochecksums variable, causes build + failures + Reported by: Guido Falsi + * [e4366308e1] Guido Falsi -- res_rtp_asterisk: Protect access to + nochecksums with #ifdef + ASTERISK-28827: res_rtp_asterisk: Loop when receive buffer is flushed by a + received packet that is also in receive buffer with NACK + Reported by: nappsoft + * [d50fd0acc0] Pirmin Walthert -- res_rtp_asterisk: Resolve loop when + receive buffer is flushed + ASTERISK-28826: res_rtp_asterisk: Duplicate seqnos being added to send + buffer with NACK + Reported by: nappsoft + * [ca032d1e2e] Pirmin Walthert -- res_rtp_asterisk: Free payload when + error on insertion to data buffer + ASTERISK-28812: First DTMF is not get + Reported by: Bernard Merindol + * [7db03e12a7] Bernard Merindol -- res_rtp_asterisk.c: Check for first + DTMF having timestamp set to 0 + ASTERISK-28809: [patch] res_rtp_asterisk: Avoid absolute value on unsigned + subtraction. + Reported by: Alexander Traud + * [1ef1b1b0c2] Alexander Traud -- res_rtp_asterisk: Avoid absolute value + on unsigned subtraction. + ASTERISK-28773: Incorrect Sender SSRC in RTCP when p2p rtp bridge is + active + Reported by: Torrey Searle + * [a1dba820cf] Torrey Searle -- res_rtp_asterisk: Send correct sender + SSRC when p2p bridge in use + ASTERISK-28769: DTLS Handshake Fails to Occur if ice_support is enabled + but not used + Reported by: Torrey Searle + * [14ba1806f3] Torrey Searle -- res_pjsip_sdp_rtp: Don't wait for ICE if + not negotiated + ASTERISK-28759: A non negotiated rtp frame causes call disconnection when + there is a SSRC change + Reported by: Paulo Vicentini + * [ed2a7e3eaf] Paulo Vicentini -- chan_pjsip: Check audio frame when + remote SSRC changes. + ASTERISK-28764: res_rtp_asterisk: Improve NACK support and seqno handling + Reported by: Joshua C. Colp + * [87fda066ea] Joshua C. Colp -- res_rtp_asterisk: Improve video + performance in certain networks. + ASTERISK-28716: ICE: pjnath shouldn't wait for ICE to complete before + allowing sending + Reported by: Benjamin Keith Ford + * [168637cc0c] Ben Ford -- RTP/ICE: Send on first valid pair. + ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete + dtls/srtp setup + Reported by: Kevin Harwell + * [3865b3fd6a] Kevin Harwell -- res_rtp_asterisk: bad audio (static) due + to incomplete dtls/srtp setup + ASTERISK-28576: res_rtp_asterisk: ICE Completion Crash when sent packet + length doesn't match + Reported by: Joshua Elson + * [02129ad4d0] Joshua Colp -- res_rtp_asterisk: Always return provided + DTLS packet length. + + Category: Resources/res_sorcery_memory_cache + + ASTERISK-28942: res_sorcery_memory_cache: Individual object expiration + behaves unexpectedly with full backend caching + Reported by: Joshua C. Colp + * [a143c3a7b7] Joshua C. Colp -- res_sorcery_memory_cache: Disallow + per-object expire with full backend. + + Category: Resources/res_speech + + ASTERISK-29040: res_speech: Assertion on format + Reported by: Nickolay V. Shmyrev + * [0319e0b07f] Nickolay Shmyrev -- res_speech: Bump reference on format + object + + Category: Resources/res_srtp + + ASTERISK-28903: res_srtp: Answered Crypto Suite might be wrong in + SDP/SDES. + Reported by: Alexander Traud + * [4de0e50c32] Alexander Traud -- res_srtp: Set all possible flags while + selecting the Crypto Suite. + ASTERISK-22920: Crash while Forwarding from TLS extension with CHANNEL + args secure_bridge_media and secure_bridge_signaling + Reported by: Shlomi Gutman + * [29070b61f7] Alexander Traud -- core_local: Local calls are always + secure. + + Category: Resources/res_stasis + + ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info + Reported by: sungtae kim + * [2e32b56bdb] sungtae kim -- stasis_bridge.c: Fixed wrong video_mode + shown + ASTERISK-28423: ARI causes STASIS Deadlock + Reported by: Ross Beer + * [cce2b0da95] Kevin Harwell -- stasis/app: don't lock an app before a + call to send + * [990a91b44a] George Joseph -- stasis: Don't hold app_registry and + session locks unnecessarily + ASTERISK-28633: stasis bridge topic leak + Reported by: Joeran Vinzens + * [1c9ddad4db] George Joseph -- stasis.c: Use correct topic name in + stasis_topic_pool_delete_topic + + Category: Resources/res_stasis_playback + + ASTERISK-28713: res_stasis_playback: Error building JSON + Reported by: Sébastien Duthil + * [31dc904380] Sean Bright -- res_stasis_playback: Prevent media_index + from going out of bounds + + Category: Resources/res_statsd + + ASTERISK-24484: Update documentation for statsd module - usage + requirements unclear + Reported by: Dan Jenkins + * [c376e9f8a8] Sean Bright -- res_statsd: Document that res_statsd does + nothing on its own + + Category: Tests/General + + ASTERISK-28808: [patch] test_stasis: Avoid always true warning with clang. + Reported by: Alexander Traud + * [bb28ed0d1b] Alexander Traud -- test_stasis: Avoid always true warning + with clang. + + Category: Tests/testsuite + + ASTERISK-27717: [patch] BuildSystem: In NetBSD, the Python Programming + Language is python-2.7. + Reported by: Alexander Traud + * [610e058189] Alexander Traud -- BuildSystem: Search for Python/C API + when possibly needed only. + ASTERISK-28789: test_utils: incorrectly printing error 'declined to load' + Reported by: Alexander Traud + * [fc07eeaba1] Alexander Traud -- test_utils: Avoid incorrect error + message on load. + + Category: Utilities/General + + ASTERISK-28685: check_expr2: linking (when hardening) and cross-compiling + troubles + Reported by: Sebastian Kemper + * [b7fbb9c41f] Sebastian Kemper -- check_expr2: fix + cross-compile/hardening issues + + Category: pjproject/pjsip + + ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered + INVITE when NAT is active (UDP transport with external_media_address) + Reported by: Michael Neuhauser + * [6482ab5bea] Michael Neuhauser -- pjproject: clone sdp to protect + against (nat) modifications + ASTERISK-28929: pjproject_bundled: Honor --without-pjproject. + Reported by: Alexander Traud + * [0a4dffe6f8] Alexander Traud -- pjproject_bundled: Honor + --without-pjproject. + ASTERISK-28794: res_pjsip: Crash when escaping during URI printing + Reported by: nappsoft + * [9c2871edf4] Joshua C. Colp -- res_pjsip: Use correct pool for storing + the contact_user value. + ASTERISK-28859: pjsip: Increase maximum candidate count + Reported by: Joshua C. Colp + * [3078a00a6d] Joshua C. Colp -- pjsip: Increase maximum ICE candidate + count. + ASTERISK-28811: Crash occurs when fax session switches from T.38 to audio + Reported by: Alexey Vasilyev + * [e56f4de7e6] Joshua C. Colp -- fax: Fix crashes in PJSIP + re-negotiation scenarios. + ASTERISK-28837: pjproject_bundled: Honor --without-pjproject. + Reported by: Alexander Traud + * [966acc6251] Alexander Traud -- pjproject_bundled: Honor + --without-pjproject. + ASTERISK-28758: pjsip startup errors when using "with-ssl" configure + option + Reported by: Patrick Wakano + * [3431949a52] Alexander Traud -- pjproject_bundled: Repair ./configure + --with-ssl without ARG. + ASTERISK-26955: pjsip: SIP Packets with Via "received=" Containing IPv6 + Address Delimited by "[]" Rejected + Reported by: Peter Sokolov + * [9d9bde76a9] Sean Bright -- pjproject_bundled: Allow brackets in via + parameters + ASTERISK-28574: pjproject fails to build on 16.6.0, works on 16.5 + Reported by: Niklas Larsson + * [5d9f9f4871] George Joseph -- pjproject_bundled: Replace earlier + reverts with official fixes. + ASTERISK-28509: PJSIP cnonce generated on Linux contains 36 characters, + NEC only supports up to 32 characters + Reported by: Dan Cropp + * [0844d6b127] Dan Cropp -- pjproject: Configurable setting for cnonce + to include hyphens or not + + Improvement + + Category: Addons/chan_mobile + + ASTERISK-28832: chan_mobile creates PCMA streams that make some VoIP + clients crash or not render received audio + Reported by: Peter Turczak + * [3303defd3f] Peter Turczak -- chan_mobile: Add smoother to make + SIP/RTP endpoints happy. + + Category: Applications/NewFeature + + ASTERISK-28484: Add AudioSocket support + Reported by: Seán C. McCord + * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, + and ARI support. + + Category: Applications/app_confbridge + + ASTERISK-28658: app_confbridge: Add support for setting maximum sample + rate + Reported by: Joshua C. Colp + * [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying + maximum sample rate. + + Category: Applications/app_mixmonitor + + ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name + Extension In MixMonitor + Reported by: xrobau + * [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to + correct value when wav49 is used + + Category: Applications/app_page + + ASTERISK-27946: dial (API): Storage of dialed target uses + AST_MAX_EXTENSION when it shouldn't + Reported by: Joshua Elson + * [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character + limitation + + Category: Applications/app_voicemail + + ASTERISK-28567: Problem with ASTERISK-20207: Asterisk should clear out any + .lock files in the voice mail directory on startup. + Reported by: Michael + * [7362647e2f] Sean Bright -- Revert "app_voicemail: Cleanup stale lock + files on module load" + + Category: Applications/app_voicemail/ODBC + + ASTERISK-22192: [patch] Allow voicemail forwards with ODBC backend when + format differs from attachfmt column + Reported by: cmaj + * [2d67dbfef5] cmaj -- app_voicemail.c: Support multiple file formats + for forwarded messages. + + Category: Bridges/bridge_native_rtp + + ASTERISK-28733: stream: Add support for adding/removing streams during + SFU/calls + Reported by: Joshua C. Colp + * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for + adding/removing streams. + + Category: Bridges/bridge_simple + + ASTERISK-28733: stream: Add support for adding/removing streams during + SFU/calls + Reported by: Joshua C. Colp + * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for + adding/removing streams. + + Category: Bridges/bridge_softmix + + ASTERISK-28733: stream: Add support for adding/removing streams during + SFU/calls + Reported by: Joshua C. Colp + * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for + adding/removing streams. + ASTERISK-28658: app_confbridge: Add support for setting maximum sample + rate + Reported by: Joshua C. Colp + * [89b7144fbd] Joshua C. Colp -- confbridge: Add support for specifying + maximum sample rate. + + Category: Channels/NewFeature + + ASTERISK-28484: Add AudioSocket support + Reported by: Seán C. McCord + * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, + and ARI support. + + Category: Channels/chan_pjsip + + ASTERISK-28638: Simplify dialplan for Dial, Page, and ChanIsAvail + Reported by: cmaj + * [fe3cce816c] Richard Mudgett -- app_chanisavail.c: Simplify dialplan + using ChanIsAvail. + * [abcb4ab321] Richard Mudgett -- app_dial.c: Simplify dialplan using + Dial. + * [d86a6ac5ce] Richard Mudgett -- app_page.c: Simplify dialplan using + Page. + + Category: Contrib/General + + ASTERISK-28726: install_prereq script uses the interactive mode when + installing aptitude + Reported by: Sylvain Afchain + * [0c02d0a450] Sylvain Afchain -- install_prereq: Install aptitude + non-interactively + + Category: Core/CodecInterface + + ASTERISK-28512: Add pass-through support for H.265 (HEVC) codec + Reported by: Florian Floimair + * [c18983207d] Florian Floimair -- core: Add H.265/HEVC passthrough + support + + Category: Core/DNS + + ASTERISK-28853: Missing include on FreeBSD + Reported by: Guido Falsi + * [97494d8984] Guido Falsi -- core/dns: Add system include required on + FreeBSD + + Category: Core/Dial + + ASTERISK-27946: dial (API): Storage of dialed target uses + AST_MAX_EXTENSION when it shouldn't + Reported by: Joshua Elson + * [dbddb6725d] sungtae kim -- dial.c: Removed dial string 80 character + limitation + + Category: Core/HTTP + + ASTERISK-28750: TLS/SSL Key too small error + Reported by: Martin Zeh + * [7f2d56fc8c] Sean Bright -- tcptls.c: Log more informative OpenSSL + errors + ASTERISK-28710: Should be able to disable the /httpstatus URI in the + built-in HTTP server + Reported by: Sean Bright + * [0dce6f746b] Sean Bright -- http: Add ability to disable /httpstatus + URI + + Category: Core/ManagerInterface + + ASTERISK-28945: AMI SendText - add Content-Type parameter + Reported by: Kevin Harwell + * [cfed0ea033] Kevin Harwell -- manager - Add Content-Type parameter to + the SendText action + + Category: Core/Streams + + ASTERISK-28733: stream: Add support for adding/removing streams during + SFU/calls + Reported by: Joshua C. Colp + * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for + adding/removing streams. + + Category: Documentation + + ASTERISK-24798: Documentation - Clarify That Format Is Set By File Name + Extension In MixMonitor + Reported by: xrobau + * [ddfb60ac2c] Sean Bright -- app_mixmonitor: Set MIXMONITOR_FILENAME to + correct value when wav49 is used + ASTERISK-28673: GET FULL VARIABLE documentation clarification + Reported by: Jonathan Harris + * [7d94bdde9d] Sean Bright -- res_agi: Improve GET FULL VARIABLE + documentation + ASTERISK-28586: Typo in README-SERIOUSLY.bestpractices.md + Reported by: Sam Banks + * [0dc7e29dd8] Sean Bright -- README-SERIOUSLY.bestpractices.md: Speling + correetions. + + Category: Functions/func_volume + + ASTERISK-28813: func_volume: Allow decimal numbers as parameter to improve + granularity + Reported by: Jean Aunis - Prescom + * [de66713fd5] Jean Aunis -- func_volume: Accept decimal number as + argument + + Category: Resources/NewFeature + + ASTERISK-28484: Add AudioSocket support + Reported by: Seán C. McCord + * [163efbd724] Seán C McCord -- feat: AudioSocket channel, application, + and ARI support. + + Category: Resources/res_ari_bridges + + ASTERISK-28629: [patch] Add an "inhibitCOLP" flag to the bridges REST API + Reported by: Jean Aunis - Prescom + * [034ac357ad] Jean Aunis -- ARI: Ability to inhibit COLP frames when + adding channels to a bridge + + Category: Resources/res_ari_channels + + ASTERISK-28896: ari: Add support for specifying variables on channel + create + Reported by: Joshua C. Colp + * [15cbff9d54] Joshua C. Colp -- ari: Allow variables to be set on + channel create. + + Category: Resources/res_http_websocket + + ASTERISK-28958: Continue reading string when ping received by websocket + Reported by: Nickolay V. Shmyrev + * [7163efd934] Nickolay Shmyrev -- res_http_websocket.c: Continue + reading after ping/pong + ASTERISK-28949: res_http_websocket: Add masking to websocket client + Reported by: Moises Silva + * [9445dac43b] Moises Silva -- res_http_websocket: Add payload masking + to the websocket client + + Category: Resources/res_pjsip + + ASTERISK-28959: res_pjsip: Added option for disable rport parameter set + Reported by: sungtae kim + * [81b5e4a73f] sungtae kim -- res_pjsip.c: Added disable_rport option + for pjsip.conf + ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option + Reported by: Kevin Harwell + * [2ee455958e] George Joseph -- codec_negotiation: Implement + outgoing_call_offer_pref + ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option + Reported by: Kevin Harwell + * [06dada3f01] Kevin Harwell -- codec negotiation: add + incoming_call_offer_prefs option + ASTERISK-28542: [patch] add the ability for asterisk to generate on-hold + re-invites + Reported by: Torrey Searle + * [b43cdc7f1e] Torrey Searle -- channel/chan_pjsip: add dialplan + function for music on hold + + Category: Resources/res_pjsip_logger + + ASTERISK-28895: res_pjsip_logger: Add tons'o'functionality + Reported by: Joshua C. Colp + * [a7aaee70c6] Joshua C. Colp -- res_pjsip_logger: Expand functionality + to improve logging. + + Category: Resources/res_pjsip_outbound_registration + + ASTERISK-28602: res_pjsip_outbound_registration: Maximum retries reached + Reported by: Daniel + * [e73eba85c1] Joshua Colp -- res_pjsip_outbound_registration: Extend + documentation for "max_retries". + + Category: Resources/res_pjsip_sdp_rtp + + ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option + Reported by: Kevin Harwell + * [2ee455958e] George Joseph -- codec_negotiation: Implement + outgoing_call_offer_pref + ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option + Reported by: Kevin Harwell + * [06dada3f01] Kevin Harwell -- codec negotiation: add + incoming_call_offer_prefs option + ASTERISK-28733: stream: Add support for adding/removing streams during + SFU/calls + Reported by: Joshua C. Colp + * [5a5be92b79] Joshua C. Colp -- bridging: Add better support for + adding/removing streams. + + Category: Resources/res_pjsip_session + + ASTERISK-28777: Codec Negotiation: add outgoing_call_offer_prefs option + Reported by: Kevin Harwell + * [2ee455958e] George Joseph -- codec_negotiation: Implement + outgoing_call_offer_pref + ASTERISK-28782: Add support for Content-Disposition header in multi-part + INVITES + Reported by: Torrey Searle + * [e12244153a] Torrey Searle -- res_pjsip_session: implement processing + of Content-Disposition + ASTERISK-28787: res_pjsip_session: Decide more intelligently when to add + video + Reported by: Joshua C. Colp + * [21e9051461] Joshua C. Colp -- res_pjsip_session: Apply intention + behind requested formats. + ASTERISK-28756: Codec Negotiation: add incoming_call_offer_pref option + Reported by: Kevin Harwell + * [06dada3f01] Kevin Harwell -- codec negotiation: add + incoming_call_offer_prefs option + + Category: Third-Party/pjproject + + ASTERISK-28866: third-party/pjproject/configure.m4 contains bashisms + Reported by: Guido Falsi + * [c831f03273] Guido Falsi -- pjproject: Remove bashism from + configure.m4 script + + Category: pjproject/pjsip + + ASTERISK-28899: Upgrade Asterisk to bundled pjproject 2.10 + Reported by: Kevin Harwell + * [415b55af5a] Kevin Harwell -- pjproject: Upgrade bundled version to + pjproject 2.10 + ASTERISK-28879: pjproject has race conditions in it's build system + Reported by: Guido Falsi + * [801d570f6e] Guido Falsi -- pjproject: Fix race condition when + building with parallel make + + ---------------------------------------------------------------------- + + Open Issues + + [Back to Top] + + This is a list of all open issues from the issue tracker that were + referenced by changes that went into this release. + + Bug + + Category: Channels/chan_sip/TCP-TLS + + ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server. + Reported by: Alexander Traud + * [da9554d925] Alexander Traud -- chan_sip: TCP/TLS client without + server. + + Category: Core/Configuration + + ASTERISK-28719: Cannot remove defaultrule from queue using realtime queues + Reported by: EDV O-TON + * [eb9252ea27] Sean Bright -- res_config_odbc: Preserve empty strings + returned by the database + + ---------------------------------------------------------------------- + + Commits Not Associated with an Issue + + [Back to Top] + + This is a list of all changes that went into this release that did not + reference a JIRA issue. + + +------------------------------------------------------------------------+ + | Revision | Author | Summary | + |------------+-------------------+---------------------------------------| + | 6fd94258f8 | Asterisk | Update for 18.0.0-rc2 | + | | Development Team | | + |------------+-------------------+---------------------------------------| + | 704cb88799 | Asterisk | Update for 18.0.0-rc1 | + | | Development Team | | + |------------+-------------------+---------------------------------------| + | f589985840 | Asterisk | Update CHANGES and UPGRADE.txt for | + | | Development Team | 18.0.0 | + |------------+-------------------+---------------------------------------| + | | | res_pjsip_session: Deferred re-INVITE | + | 5a49757e40 | Patrick Verzele | without SDP send a=sendrecv instead | + | | | of a=sendonly | + |------------+-------------------+---------------------------------------| + | ec03909831 | Kevin Harwell | conversions: Add string to signed | + | | | integer conversion functions | + |------------+-------------------+---------------------------------------| + | e32815dddb | George Joseph | ast_coredumper: Fix issues with | + | | | naming | + |------------+-------------------+---------------------------------------| + | 9ed1b1452d | Alexander Traud | sip_nat_settings: Update script for | + | | | latest Linux. | + |------------+-------------------+---------------------------------------| + | 217449a1e5 | Alexander Traud | samples: Fix keep_alive_interval | + | | | default in pjsip.conf. | + |------------+-------------------+---------------------------------------| + | 5a8cacb93d | George Joseph | logger.c: Added a new log formatter | + | | | called "plain" | + |------------+-------------------+---------------------------------------| + | | | res_musiconhold.c: Use | + | 5dfeeba623 | Sean Bright | ast_file_read_dir to scan MoH | + | | | directory | + |------------+-------------------+---------------------------------------| + | c4c72d55a2 | George Joseph | scope_trace: Added debug messages and | + | | | added additional macros | + |------------+-------------------+---------------------------------------| + | d26ab7f8f9 | George Joseph | stream.c: Added 2 more debugging | + | | | utils and added pos to stream string | + |------------+-------------------+---------------------------------------| + | 6faf76308d | George Joseph | ACN: Changes specific to the core | + |------------+-------------------+---------------------------------------| + | a3d87f78ed | Joshua C. Colp | res_pjsip: Fix codec preference | + | | | defaults. | + |------------+-------------------+---------------------------------------| + | | | vector.h: Fix implementation of | + | da8a617dc9 | Sean Bright | AST_VECTOR_COMPACT() for empty | + | | | vectors | + |------------+-------------------+---------------------------------------| + | 769a9611e7 | Ben Ford | utils.c: NULL terminate | + | | | ast_base64decode_string. | + |------------+-------------------+---------------------------------------| + | 802aa97fa0 | George Joseph | ACN: Configuration renaming for pjsip | + | | | endpoint | + |------------+-------------------+---------------------------------------| + | de23cb4002 | Ben Ford | res_stir_shaken: Fix memory | + | | | allocation error in curl.c | + |------------+-------------------+---------------------------------------| + | 71446b68fc | George Joseph | res_pjsip_session: Ensure reused | + | | | streams have correct bundle group | + |------------+-------------------+---------------------------------------| + | d9ae902f52 | Sean Bright | utf8.c: Add UTF-8 validation and | + | | | utility functions | + |------------+-------------------+---------------------------------------| + | 9022f35f09 | Sean Bright | vector.h: Add AST_VECTOR_SORT() | + |------------+-------------------+---------------------------------------| + | a678dafac8 | George Joseph | CI: Force publishAsteriskDocs to use | + | | | python2 | + |------------+-------------------+---------------------------------------| + | af70bbb13a | Joshua C. Colp | websocket / pjsip: Increase maximum | + | | | packet size. | + |------------+-------------------+---------------------------------------| + | 8d15f72721 | Joshua C. Colp | pjsip: Include timer patch to prevent | + | | | cancelling timer 0. | + |------------+-------------------+---------------------------------------| + | 3330764213 | George Joseph | Update .gitreview defaultbranch to 18 | + |------------+-------------------+---------------------------------------| + | 1f5e6805bf | Asterisk | Update CHANGES and UPGRADE.txt for | + | | Development Team | 18.0.0 | + |------------+-------------------+---------------------------------------| + | 5fbed5af24 | Ben Ford | res_stir_shaken: Add stir_shaken | + | | | option and general improvements. | + |------------+-------------------+---------------------------------------| + | e88beedd08 | George Joseph | res_pjsip_session: Fix segv in | + | | | session_on_rx_response | + |------------+-------------------+---------------------------------------| + | 9bd1d686a1 | George Joseph | ACN: Add tracing to existing code | + |------------+-------------------+---------------------------------------| + | 2d22e34206 | George Joseph | ACN: res_pjsip endpoint options | + |------------+-------------------+---------------------------------------| + | d093e44b1e | George Joseph | frame.c: Make debugging easier | + |------------+-------------------+---------------------------------------| + | 955b7b4fdb | George Joseph | Scope Trace: Make it easier to trace | + | | | through synchronous tasks | + |------------+-------------------+---------------------------------------| + | 8d1064eaaf | George Joseph | Streams: Add features for Advanced | + | | | Codec Negotiation | + |------------+-------------------+---------------------------------------| + | 7440fd0397 | George Joseph | Scope Trace: Add some new tracing | + | | | macros and an ast_str helper | + |------------+-------------------+---------------------------------------| + | 1274117102 | Ben Ford | res_stir_shaken: Add outbound INVITE | + | | | support. | + |------------+-------------------+---------------------------------------| + | | | res_pjsip: Include | + | f1cfd54976 | Walter Doekes | instead of internal | + | | | "pjsua-lib/pjsua.h" | + |------------+-------------------+---------------------------------------| + | b9f42a717e | George Joseph | app_confbridge: Plug ref leak of | + | | | bridge channel with send_events | + |------------+-------------------+---------------------------------------| + | 3d1bf3c537 | Kevin Harwell | Compiler fixes for gcc 10 | + |------------+-------------------+---------------------------------------| + | 559fa0e89c | Ben Ford | cli.c: Fix compiler error. | + |------------+-------------------+---------------------------------------| + | 3927f79cb5 | Ben Ford | res_stir_shaken: Add inbound INVITE | + | | | support. | + |------------+-------------------+---------------------------------------| + | 1fcb6b1b21 | Joshua C. Colp | bridge_channel: Don't queue unmapped | + | | | frames. | + |------------+-------------------+---------------------------------------| + | ca3c22c5f1 | George Joseph | Scope Tracing: A new facility for | + | | | tracing scope enter/exit | + |------------+-------------------+---------------------------------------| + | ec7890d7c6 | Joshua C. Colp | res_sorcery_config: Always reload | + | | | configuration on errors. | + |------------+-------------------+---------------------------------------| + | f506cc4896 | Ben Ford | res_stir_shaken: Add unit tests for | + | | | signing and verification. | + |------------+-------------------+---------------------------------------| + | e29df34de0 | Ben Ford | res_stir_shaken: Added dialplan | + | | | function and API call. | + |------------+-------------------+---------------------------------------| + | 44e5dd288b | Jaco Kroon | Remove #include | + |------------+-------------------+---------------------------------------| + | 1cfd30bd8a | Joshua C. Colp | res_stir_shaken: Use ast_asprintf for | + | | | creating file path. | + |------------+-------------------+---------------------------------------| + | 9acf840f7c | Ben Ford | res_stir_shaken: Implemented | + | | | signature verification. | + |------------+-------------------+---------------------------------------| + | 7baf2c4bf1 | George Joseph | app_voicemail: Add workaround for a | + | | | gcc 10 issue with -Wrestrict | + |------------+-------------------+---------------------------------------| + | 4ef5ba58f5 | Alexander Traud | BuildSystem: Only if found LibPRI, | + | | | check its optional parts. | + |------------+-------------------+---------------------------------------| + | ef580f96e7 | Alexander Traud | BuildSystem: Only if found external | + | | | PJProject, check its optional parts. | + |------------+-------------------+---------------------------------------| + | 611529fa52 | Alexander Traud | res_stir_shaken: Do not build without | + | | | OpenSSL. | + |------------+-------------------+---------------------------------------| + | | | res_audiosocket: Avoid | + | 27de0c9700 | Alexander Traud | Sometimes-uninitialized Warning with | + | | | Clang. | + |------------+-------------------+---------------------------------------| + | 2b80e5f5da | Jaco Kroon | res_rtp_asterisk: iterate all local | + | | | addresses looking to populate ICE. | + |------------+-------------------+---------------------------------------| + | 1cf569ba2b | Jaco Kroon | res_pjsip: document legal dtls_verify | + | | | endpoint options. | + |------------+-------------------+---------------------------------------| + | 52ecbbd014 | Alexander Traud | _pjsua: Build even with Clang. | + |------------+-------------------+---------------------------------------| + | ee1c7f465b | Alexander Traud | res_rtp_asterisk: Build without | + | | | PJProject. | + |------------+-------------------+---------------------------------------| + | | | Revert "res_config_odbc: Preserve | + | 60925c68e8 | Sean Bright | empty strings returned by the | + | | | database" | + |------------+-------------------+---------------------------------------| + | c5f3836bcc | Jaco Kroon | main/backtrace: binutils-2.34 fix. | + |------------+-------------------+---------------------------------------| + | 7ba6d43083 | George Joseph | test_res_pjsip_session_caps: Create | + | | | unit test | + |------------+-------------------+---------------------------------------| + | 57a457c26c | Ben Ford | res_stir_shaken: Implemented signing | + | | | of JSON payload. | + |------------+-------------------+---------------------------------------| + | d32e559e8a | Jaco Kroon | acl: implement a centralized ACL | + | | | output mechanism for HAs and ACLs. | + |------------+-------------------+---------------------------------------| + | 1b6c58896f | Joshua C. Colp | chan_sip: Send 403 when ACL fails. | + |------------+-------------------+---------------------------------------| + | 3ed80fc57b | Joshua C. Colp | CHANGES: Change md file extension to | + | | | txt. | + |------------+-------------------+---------------------------------------| + | 26713dc88b | Kevin Harwell | ast_coredumper: add Asterisk | + | | | information dump | + |------------+-------------------+---------------------------------------| + | 6f731f153b | Jaco Kroon | netsock2: compile fixes. | + |------------+-------------------+---------------------------------------| + | 211bb8a79c | Ben Ford | res_stir_shaken: Initial commit and | + | | | reading private key. | + |------------+-------------------+---------------------------------------| + | a699e016dd | Jaco Kroon | build: enable building with uClibc | + |------------+-------------------+---------------------------------------| + | f824cd6a13 | Jaco Kroon | build: search from newest to oldest | + | | | for gmime. | + |------------+-------------------+---------------------------------------| + | 82c3939c38 | Jaco Kroon | res_rtp_asterisk: implement ACL | + | | | mechanism for ICE and STUN addresses. | + |------------+-------------------+---------------------------------------| + | 2ad64e97c0 | Jaco Kroon | Update main/backtrace.c to deal with | + | | | changes in binutils 2.34. | + |------------+-------------------+---------------------------------------| + | 49cf84578e | Sean Bright | chan_vpb: Fix 'catching polymorphic | + | | | type ... by value' error | + |------------+-------------------+---------------------------------------| + | d68f940f6e | Sean Bright | dns_txt: Add TXT record parsing | + | | | support | + |------------+-------------------+---------------------------------------| + | 00a7e4b51d | George Joseph | CI: Create generic jenkinsfile | + |------------+-------------------+---------------------------------------| + | e089779908 | Rodrigo RamÃrez | res_rtp_asterisk: Add 'rtp show | + | | Norambuena | settings' cli command | + |------------+-------------------+---------------------------------------| + | 680e6b9774 | Walter Doekes | app_queue: Refactor odd placement of | + | | | if's around say_position | + |------------+-------------------+---------------------------------------| + | 1e1651b4f4 | Kevin Harwell | format_cap: make function parameters | + | | | 'const' | + |------------+-------------------+---------------------------------------| + | 0b5c6fddf1 | Walter Doekes | say: Remove unused "plural" option | + | | | from main/say | + |------------+-------------------+---------------------------------------| + | 5cd7230f3c | Jaco Kroon | addons/res_config_mysql: silense | + | | | warnings about printf format errors. | + |------------+-------------------+---------------------------------------| + | de6919f339 | Sean Bright | ast_tls_cert: Allow private key size | + | | | to be set on command line | + |------------+-------------------+---------------------------------------| + | 8dcdce42a9 | Sean Bright | app_mixmonitor: Turn on | + | | | synchronization by default | + |------------+-------------------+---------------------------------------| + | 0f6ee98c3f | Joshua C. Colp | stasis: Use format specifier for | + | | | size_t. | + |------------+-------------------+---------------------------------------| + | 1e037ebb97 | Sean Bright | func_odbc: Prevent snprintf() | + | | | truncation warning | + |------------+-------------------+---------------------------------------| + | a72caa041f | George Joseph | doc: Fix CHANGES entries to have .txt | + | | | suffix and update READMEs | + |------------+-------------------+---------------------------------------| + | 1b53d329ac | Joshua C. Colp | res_rtp_asterisk: Don't produce | + | | | transport-cc if no packets. | + |------------+-------------------+---------------------------------------| + | b76ab5e5c9 | George Joseph | message.c: Add option to suppress the | + | | | Message channel AMI and ARI events | + |------------+-------------------+---------------------------------------| + | 113d05e504 | Walter Doekes | chan_sip: Clarify in sample docs how | + | | | directmediapermit/-acl should be used | + |------------+-------------------+---------------------------------------| + | 262221f4d9 | Sean Bright | func_odbc.conf.sample: Add example | + | | | lookup | + |------------+-------------------+---------------------------------------| + | f09cf4da44 | Sean Bright | app_voicemail: Remove MessageExists | + | | | and MESSAGE_EXISTS() | + |------------+-------------------+---------------------------------------| + | 5cbf47714a | Sean Bright | app_voicemail, say: Fix various | + | | | leading whitespace problems | + |------------+-------------------+---------------------------------------| + | | | netsock2: | + | 3bc8b36537 | Jaco Kroon | ast_addressfamily_to_sockaddrsize and | + | | | ast_sockaddr_from_sockaddr. | + |------------+-------------------+---------------------------------------| + | 00a7432156 | Kevin Harwell | app_agent_pool: Update XML docs for | + | | | AgentLogin | + |------------+-------------------+---------------------------------------| + | 19069f7db7 | Richard Mudgett | app_bridgeaddchan.c: Make BridgeAdd | + | | | be more like Bridge | + |------------+-------------------+---------------------------------------| + | 0376f2bba9 | Richard Mudgett | features.c: Make Bridge application | + | | | tolerate unspecified channel. | + |------------+-------------------+---------------------------------------| + | 0d1f3d9bf3 | Richard Mudgett | app_chanspy.c: Reduce log message | + | | | level from notice to verbose. | + |------------+-------------------+---------------------------------------| + | a457947198 | Richard Mudgett | app_softhangup.c: Reduce unnecessary | + | | | warning to verbose message. | + |------------+-------------------+---------------------------------------| + | fc99ac8c9a | Sean Bright | db: Initialize condition primitive | + | | | before use | + |------------+-------------------+---------------------------------------| + | 32160cb456 | Jaco Kroon | ACL: ast_apply_acl_nolog - identical | + | | | to ast_apply_acl but without logging. | + |------------+-------------------+---------------------------------------| + | d0b198b330 | Joshua Colp | Revert "PJSIP_CONTACT: add missing | + | | | argument documentation" | + |------------+-------------------+---------------------------------------| + | | | res_pjsip_registrar.c: Prevent | + | 0183e2bc67 | Sean Bright | possible buffer overflow with domain | + | | | aliases | + |------------+-------------------+---------------------------------------| + | fd823225a6 | Thomas Arimont | channel.c: Resolve issue with | + | | | receiving SIP INFO packets for DTMF | + |------------+-------------------+---------------------------------------| + | 366da90f74 | George Joseph | CI: Turn off shallow cloning | + | | | altogether | + |------------+-------------------+---------------------------------------| + | 91c3b5b09d | Sean Bright | media_cache.c: Various CLI | + | | | improvements | + |------------+-------------------+---------------------------------------| + | 48161dfc71 | Rodrigo RamÃrez | queue_log: Add alembic script for | + | | Norambuena | generate db table for queue_log | + |------------+-------------------+---------------------------------------| + | 2a6a2800e7 | George Joseph | CI: Fix missing script block in | + | | | jenkinsfiles | + |------------+-------------------+---------------------------------------| + | 4abb54b2e4 | George Joseph | CI: Fix missing script block in | + | | | jenkinsfiles | + |------------+-------------------+---------------------------------------| + | e8e1314fcb | George Joseph | CI: Increase clone depth and do | + | | | better cleanup | + |------------+-------------------+---------------------------------------| + | a5fa0d662e | Sean Bright | res_pjsip_registrar: Fix uninitlized | + | | | variable warning | + |------------+-------------------+---------------------------------------| + | f2d5ed54ea | Alexei Gradinari | serializer: set high/low alert levels | + | | | on whole pool | + |------------+-------------------+---------------------------------------| + | bdd785d31c | Kevin Harwell | various files - fix some alerts | + | | | raised by lgtm code analysis | + |------------+-------------------+---------------------------------------| + | 0e3b397812 | Kevin Harwell | res_pjsip_session: initialize | + | | | pending's topology to endpoint's | + |------------+-------------------+---------------------------------------| + | 8a1f30af04 | Corey Farrell | core: Improve MALLOC_DEBUG for | + | | | frames. | + |------------+-------------------+---------------------------------------| + | d71d0f9489 | George Joseph | ExternalMedia: Change return object | + | | | from ExternalMedia to Channel | + |------------+-------------------+---------------------------------------| + | 6e907ae5d4 | Joshua Colp | res_rtp_asterisk: Remove a log | + | | | message that slipped in. | + |------------+-------------------+---------------------------------------| + | a60d2e905c | Joshua Colp | test_res_rtp: Enable FIR and REMB | + | | | nominal tests. | + |------------+-------------------+---------------------------------------| + | b27a5183da | Chris Savinovich | test_taskprocessor.c: Fix test | + | | | failure on Ubuntu | + |------------+-------------------+---------------------------------------| + | c0efe19cec | Kevin Harwell | serializer: move/add asterisk | + | | | serializer pool functionality | + |------------+-------------------+---------------------------------------| + | 2970a13fb8 | Kevin Harwell | res_pjsip/res_pjsip_mwi: use | + | | | centralized serializer pools | + |------------+-------------------+---------------------------------------| + | 068ed2c626 | Alexei Gradinari | res_pjsip_pubsub: add endpoint to | + | | | some warning | + |------------+-------------------+---------------------------------------| + | ba64d68273 | Jonathan Rose | basic-pbx: Bring forward queue | + | | | configuration from 13 | + |------------+-------------------+---------------------------------------| + | 4c3655ecfd | Ben Ford | taskprocessor.c: Added "like" support | + | | | to 'core show taskprocessors' | + |------------+-------------------+---------------------------------------| + | 966488ab52 | Sean Bright | res_musiconhold: Add new 'playlist' | + | | | mode | + |------------+-------------------+---------------------------------------| + | f7045cefd9 | Corey Farrell | stasis_state: Create internal | + | | | stasis_state_proxy object. | + |------------+-------------------+---------------------------------------| + | 67ba62f4e6 | Kevin Harwell | res_pjsip_pubsub: change warning to | + | | | debug | + |------------+-------------------+---------------------------------------| + | 4de1e6d0e6 | Ben Ford | taskprocessor.c: Add CLI commands to | + | | | reset taskprocessor stats. | + |------------+-------------------+---------------------------------------| + | 725e991faf | Corey Farrell | core: Add AO2_ALLOC_OPT_NO_REF_DEBUG | + | | | option. | + |------------+-------------------+---------------------------------------| + | e82f2f6e82 | George Joseph | astmm.c: Display backtrace with | + | | | memory show allocations | + |------------+-------------------+---------------------------------------| + | a4142c8437 | Corey Farrell | core: Fix ABI mismatch of | + | | | ao2_global_obj. | + |------------+-------------------+---------------------------------------| + | ca608d2575 | Corey Farrell | stasis: refcounter.py can incorrectly | + | | | report skewed objects. | + |------------+-------------------+---------------------------------------| + | 3dfbc05c53 | Corey Farrell | stasis: Fix leaks | + |------------+-------------------+---------------------------------------| + | 863fe2225f | Corey Farrell | app_voicemail: Fix module unload | + | | | leak. | + |------------+-------------------+---------------------------------------| + | 723b695ce5 | Ben Ford | res_rtp_asterisk.c: Send RTCP as | + | | | compound packets. | + |------------+-------------------+---------------------------------------| + | 0e56643d9f | Ben Ford | res_rtp: Add unit tests for RTCP | + | | | stats. | + |------------+-------------------+---------------------------------------| + | 2ae1a22e0e | George Joseph | ARI: External Media | + |------------+-------------------+---------------------------------------| + | 5fb9b23105 | George Joseph | chan_sip: Update links referenced in | + | | | deprecation notice | + |------------+-------------------+---------------------------------------| + | | | test_utils.c: Skip test | + | ed757cc7bb | Chris-Savinovich | adsi_loaded_test if module not | + | | | loaded. | + |------------+-------------------+---------------------------------------| + | 1d06a1efb3 | Igor Goncharovsky | chan_unistim: Fix code, causing all | + | | | incoming DTMF sent back to asterisk | + |------------+-------------------+---------------------------------------| + | 649003821e | Igor Goncharovsky | chan_unistim: Fix RTP port byte order | + | | | for big-endian arch | + |------------+-------------------+---------------------------------------| + | 3ef52b0b17 | Alexei Gradinari | Fix misname 'res_external_mwi' to | + | | | 'res_mwi_external' in comments. | + |------------+-------------------+---------------------------------------| + | 19045db392 | George Joseph | chan_rtp: Accept hostname as well as | + | | | ip address as destination | + |------------+-------------------+---------------------------------------| + | 9e015713cc | George Joseph | dns_core: Create new API | + | | | ast_dns_resolve_ipv6_and_ipv4 | + |------------+-------------------+---------------------------------------| + | 8da4e28a81 | George Joseph | res_ari.c: Prefer exact handler match | + | | | over wildcard | + |------------+-------------------+---------------------------------------| + | 64906c4c9b | Sean Bright | audiohook.c: Substitute silence for | + | | | unavailable audio frames | + |------------+-------------------+---------------------------------------| + | 446bac733d | George Joseph | CI: Escape backslashes in | + | | | printenv/sort/tr | + |------------+-------------------+---------------------------------------| + | be6130607d | George Joseph | CI: Add "throttle" label and | + | | | "skip_gate" capability | + |------------+-------------------+---------------------------------------| + | c01dd2a41a | George Joseph | CI: Make node labels job-specific | + |------------+-------------------+---------------------------------------| + | 9d07d5a6d6 | Sean Bright | app_voicemail: Remove extra | + | | | menuselect build options | + |------------+-------------------+---------------------------------------| + | 1f8ae708a0 | Sean Bright | res_musiconhold: Use a vector instead | + | | | of custom array allocation | + |------------+-------------------+---------------------------------------| + | 5f66fb5139 | Sean Bright | manager: Send fewer packets | + |------------+-------------------+---------------------------------------| + | 5e6e1175d5 | Asterisk | Update CHANGES and UPGRADE.txt for | + | | Development Team | 17.0.0 | + |------------+-------------------+---------------------------------------| + | 8d10028b98 | George Joseph | Update master for Asterisk 18 | + |------------+-------------------+---------------------------------------| + | 7ce9ee7f2e | Sean Bright | res_musiconhold: Use | + | | | ast_pipe_nonblock() wrapper | + |------------+-------------------+---------------------------------------| + | 8e44d823c1 | George Joseph | loader.c: Fix possible SEGV when a | + | | | module fails to register | + +------------------------------------------------------------------------+ + + ---------------------------------------------------------------------- + + Diffstat Results + + [Back to Top] + + This is a summary of the changes to the source code that went into this + release that was generated using the diffstat utility. + + asterisk-17.0.0-summary.html | 1265 -- + asterisk-17.0.0-summary.txt | 2973 ---- + b/.gitreview | 2 + b/.version | 2 + b/CHANGES | 400 + b/ChangeLog | 6317 +++++++++- + b/Makefile | 8 + b/Makefile.rules | 19 + b/README-SERIOUSLY.bestpractices.md | 4 + b/UPGRADE.txt | 120 + b/addons/cdr_mysql.c | 18 + b/addons/chan_mobile.c | 63 + b/addons/chan_ooh323.c | 7 + b/addons/ooh323c/src/decode.c | 10 + b/addons/ooh323c/src/ooSocket.c | 2 + b/addons/ooh323c/src/oochannels.c | 2 + b/addons/res_config_mysql.c | 16 + b/apps/app_agent_pool.c | 4 + b/apps/app_amd.c | 24 + b/apps/app_audiosocket.c | 240 + b/apps/app_bridgeaddchan.c | 62 + b/apps/app_cdr.c | 8 + b/apps/app_chanisavail.c | 137 + b/apps/app_chanspy.c | 3 + b/apps/app_confbridge.c | 9 + b/apps/app_dahdiras.c | 6 + b/apps/app_dial.c | 108 + b/apps/app_dictate.c | 4 + b/apps/app_fax.c | 2 + b/apps/app_followme.c | 10 + b/apps/app_getcpeid.c | 1 + b/apps/app_meetme.c | 2 + b/apps/app_minivm.c | 3 + b/apps/app_mixmonitor.c | 51 + b/apps/app_osplookup.c | 13 + b/apps/app_page.c | 30 + b/apps/app_playback.c | 9 + b/apps/app_queue.c | 201 + b/apps/app_readexten.c | 3 + b/apps/app_record.c | 3 + b/apps/app_senddtmf.c | 13 + b/apps/app_softhangup.c | 2 + b/apps/app_stack.c | 11 + b/apps/app_stream_echo.c | 2 + b/apps/app_voicemail.c | 839 - + b/apps/confbridge/conf_config_parser.c | 30 + b/apps/confbridge/confbridge_manager.c | 2 + b/apps/confbridge/include/confbridge.h | 2 + b/asterisk-18.0.0-rc2-summary.html | 15 + b/asterisk-18.0.0-rc2-summary.txt | 83 + b/bridges/bridge_native_rtp.c | 173 + b/bridges/bridge_simple.c | 200 + b/bridges/bridge_softmix.c | 281 + b/cdr/cdr_odbc.c | 2 + b/cdr/cdr_pgsql.c | 2 + b/cel/cel_pgsql.c | 2 + b/channels/Makefile | 2 + b/channels/chan_audiosocket.c | 302 + b/channels/chan_dahdi.c | 39 + b/channels/chan_dahdi.h | 18 + b/channels/chan_iax2.c | 16 + b/channels/chan_motif.c | 9 + b/channels/chan_pjsip.c | 314 + b/channels/chan_rtp.c | 19 + b/channels/chan_sip.c | 268 + b/channels/chan_unistim.c | 178 + b/channels/chan_vpb.cc | 2 + b/channels/iax2/parser.c | 18 + b/channels/pjsip/cli_commands.c | 13 + b/channels/pjsip/dialplan_functions.c | 84 + b/channels/pjsip/include/dialplan_functions.h | 25 + b/channels/sig_pri.c | 25 + b/channels/sip/include/sip.h | 1 + b/codecs/Makefile | 3 + 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b/doc/CHANGES-staging/hide_messaging_ami_events | 11 + b/doc/UPGRADE-staging/README.md | 7 + b/doc/appdocsxml.dtd | 2 + b/formats/format_g726.c | 16 + b/formats/msgsm.h | 4 + b/funcs/func_aes.c | 4 + b/funcs/func_channel.c | 8 + b/funcs/func_curl.c | 49 + b/funcs/func_env.c | 5 + b/funcs/func_jitterbuffer.c | 19 + b/funcs/func_odbc.c | 26 + b/funcs/func_pjsip_contact.c | 6 + b/funcs/func_volume.c | 12 + b/include/asterisk/abstract_jb.h | 4 + b/include/asterisk/acl.h | 69 + b/include/asterisk/app.h | 7 + b/include/asterisk/ari.h | 2 + b/include/asterisk/astobj2.h | 61 + b/include/asterisk/audiohook.h | 2 + b/include/asterisk/autoconfig.h.in | 9 + b/include/asterisk/bridge.h | 21 + b/include/asterisk/bridge_features.h | 4 + b/include/asterisk/calendar.h | 4 + b/include/asterisk/channel.h | 87 + b/include/asterisk/channel_internal.h | 5 + b/include/asterisk/config.h | 18 + b/include/asterisk/config_options.h | 2 + b/include/asterisk/conversions.h | 54 + b/include/asterisk/core_unreal.h | 19 + b/include/asterisk/dns_core.h | 22 + b/include/asterisk/dns_internal.h | 31 + b/include/asterisk/dns_txt.h | 64 + b/include/asterisk/format_cache.h | 5 + b/include/asterisk/format_cap.h | 15 + b/include/asterisk/frame.h | 40 + b/include/asterisk/http_websocket.h | 14 + b/include/asterisk/iostream.h | 14 + b/include/asterisk/logger.h | 426 + b/include/asterisk/manager.h | 2 + b/include/asterisk/max_forwards.h | 1 + b/include/asterisk/message.h | 31 + b/include/asterisk/mixmonitor.h | 5 + b/include/asterisk/netsock2.h | 42 + b/include/asterisk/options.h | 7 + b/include/asterisk/parking.h | 5 + b/include/asterisk/res_audiosocket.h | 87 + b/include/asterisk/res_fax.h | 3 + b/include/asterisk/res_pjsip.h | 106 + b/include/asterisk/res_pjsip_presence_xml.h | 5 + b/include/asterisk/res_pjsip_pubsub.h | 23 + b/include/asterisk/res_pjsip_session.h | 27 + b/include/asterisk/res_pjsip_session_caps.h | 82 + b/include/asterisk/res_stir_shaken.h | 117 + b/include/asterisk/rtp_engine.h | 111 + 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b/res/res_calendar_icalendar.c | 1 + b/res/res_config_curl.c | 7 + b/res/res_config_pgsql.c | 2 + b/res/res_corosync.c | 564 + b/res/res_fax.c | 37 + b/res/res_http_websocket.c | 97 + b/res/res_musiconhold.c | 472 + b/res/res_phoneprov.c | 6 + b/res/res_pjsip.c | 492 + b/res/res_pjsip/config_system.c | 8 + b/res/res_pjsip/config_transport.c | 17 + b/res/res_pjsip/location.c | 25 + b/res/res_pjsip/pjsip_configuration.c | 209 + b/res/res_pjsip/pjsip_message_filter.c | 40 + b/res/res_pjsip/pjsip_options.c | 7 + b/res/res_pjsip/pjsip_resolver.c | 30 + b/res/res_pjsip_acl.c | 20 + b/res/res_pjsip_caller_id.c | 3 + b/res/res_pjsip_config_wizard.c | 7 + b/res/res_pjsip_dialog_info_body_generator.c | 80 + b/res/res_pjsip_diversion.c | 7 + b/res/res_pjsip_dlg_options.c | 3 + b/res/res_pjsip_dtmf_info.c | 1 + b/res/res_pjsip_empty_info.c | 1 + b/res/res_pjsip_endpoint_identifier_ip.c | 102 + b/res/res_pjsip_history.c | 4 + b/res/res_pjsip_logger.c | 451 + b/res/res_pjsip_messaging.c | 58 + b/res/res_pjsip_mwi.c | 339 + b/res/res_pjsip_nat.c | 110 + b/res/res_pjsip_notify.c | 22 + b/res/res_pjsip_one_touch_record_info.c | 1 + b/res/res_pjsip_outbound_registration.c | 66 + b/res/res_pjsip_path.c | 1 + b/res/res_pjsip_pubsub.c | 148 + b/res/res_pjsip_refer.c | 8 + b/res/res_pjsip_registrar.c | 80 + b/res/res_pjsip_rfc3326.c | 1 + b/res/res_pjsip_sdp_rtp.c | 283 + b/res/res_pjsip_session.c | 667 - + b/res/res_pjsip_session.exports.in | 1 + b/res/res_pjsip_session/pjsip_session_caps.c | 164 + b/res/res_pjsip_stir_shaken.c | 330 + b/res/res_pjsip_t38.c | 40 + b/res/res_pjsip_transport_websocket.c | 4 + b/res/res_realtime.c | 56 + b/res/res_resolver_unbound.c | 6 + b/res/res_rtp_asterisk.c | 1100 + + b/res/res_sorcery_config.c | 14 + b/res/res_sorcery_memory_cache.c | 18 + b/res/res_speech.c | 7 + b/res/res_srtp.c | 8 + b/res/res_stasis.c | 57 + b/res/res_stasis_playback.c | 4 + b/res/res_statsd.c | 35 + b/res/res_stir_shaken.c | 1663 ++ + b/res/res_stir_shaken.exports.in | 6 + 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+ b/tests/CI/ref_debug.jenkinsfile | 9 + b/tests/CI/unittests.jenkinsfile | 9 + b/tests/CI/universal-asterisk-nongerrit.jenkinsfile | 452 + b/tests/test_conversions.c | 153 + b/tests/test_data_buffer.c | 2 + b/tests/test_locale.c | 12 + b/tests/test_res_pjsip_session_caps.c | 176 + b/tests/test_res_rtp.c | 516 + b/tests/test_scope_trace.c | 126 + b/tests/test_stasis.c | 2 + b/tests/test_stream.c | 2 + b/tests/test_strings.c | 39 + b/tests/test_taskprocessor.c | 78 + b/tests/test_utils.c | 11 + b/third-party/pjproject/Makefile | 3 + b/third-party/pjproject/configure.m4 | 199 + b/third-party/pjproject/patches/0011-sip_inv_patch.patch | 39 + b/third-party/pjproject/patches/0020-pjlib_cancel_timer_0.patch | 39 + b/third-party/pjproject/patches/0050-fix-race-parallel-build.patch | 72 + b/third-party/pjproject/patches/0060-clone-sdp-for-sip-timer-refresh-invite.patch | 28 + b/third-party/pjproject/patches/config_site.h | 4 + b/third-party/pjproject/pjproject-2.10.tar.bz2.md5 | 2 + b/third-party/versions.mak | 2 + b/utils/Makefile | 6 + b/utils/astman.c | 2 + b/utils/db1-ast/hash/ndbm.c | 3 + b/utils/db1-ast/include/db.h | 1 + b/utils/extconf.c | 10 + b/utils/muted.c | 11 + third-party/pjproject/patches/0010-ssl_sock_ossl-sip_transport_tls-Add-peer-to-error-me.patch | 157 + third-party/pjproject/patches/0020-patch_cnonce_only_digits_option.patch | 53 + third-party/pjproject/patches/0030-ssl-regression-fix.patch | 105 + third-party/pjproject/patches/0031-transport-regression-fix.patch | 187 + third-party/pjproject/pjproject-2.9.tar.bz2.md5 | 2 + 392 files changed, 29541 insertions(+), 9926 deletions(-) -- 2.47.2