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