asterisk-14.2.0-rc1
Date: 2016-11-18
<asteriskteam@digium.com>
Table of Contents
- Summary
- Contributors
- Closed Issues
- Open Issues
- Other Changes
- Diffstat
Summary
[Back to Top]This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.
The data in this summary reflects changes that have been made since the previous release, asterisk-14.1.0.
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 |
|---|
20 George Joseph
13 Joshua Colp
12 Mark Michelson
10 Matt Jordan
7 Richard Mudgett
3 Kevin Harwell
3 Alexander Traud
3 Corey Farrell
3 Tzafrir Cohen
3 Alexander Anikin
2 Sebastian Gutierrez
1 Rusty Newton
1 Alexei Gradinari
1 Pascal Cadotte Michaud
1 snuffy
1 Matt Krokosz
1 Igor Goncharovsky
1 Leandro Dardini
1 Grachev Sergey
1 Moises Silva
1 Michael Kuron
1 Michael Walton
1 Etienne Lessard
| 1 Dmitry Melekhov
| 8 Matt Jordan
5 Joshua Colp
5 Alexander Traud
4 Morten Tryfoss
4 scgm11
3 George Joseph
3 Richard Mudgett
2 Andrew Nagy
2 Gabriele Giacone <1o5g4r8o@gmail.com>
1 Rusty Newton
1 Andreas Wetzel
1 Alexei Gradinari
1 Bill Brigden
1 Sergey Grachev
1 snuffy
1 Daniele Pallastrelli
1 Kevin Harwell
1 Kayode
1 Michael Keuter
1 Dmitry Melekhov
1 Harley Peters
1 Corey Farrell
1 Leandro Dardini
1 Jonathan Harris
1 Frankie Chin
1 Badalian Vyacheslav
1 Doug Lytle
1 Michael Walton
1 Etienne Lessard
1 Tzafrir Cohen
1 John Kiniston
1 Jason
1 Florian Loyau
1 Michelle Dupuis
1 Ian Gilmour
1 Matt Krokosz
1 Mark Michelson
1 Morton Tryfoss
|
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.
Improvement
Category: Applications/app_queue
ASTERISK-26558: app_queue: add variable to know if the call is not answered after a queue
Reported by: scgm11
- [90389f6b7d] Joshua Colp -- app_queue: Add mention of 'ABANDON' variable to CHANGES.
- [abd41590d7] Sebastian Gutierrez -- app_queue: new variable set when abandoned
Category: Channels/chan_sip/General
ASTERISK-26176: chan_sip: Add AccountCode to AMI PeerEntry
Reported by: scgm11
- [1012c28437] Sebastian Gutierrez -- chan_sip: add missing account code
Category: Codecs/codec_opus
ASTERISK-26538: codec_opus: Add sample to configs/samples/codecs.conf.sample
Reported by: Kevin Harwell
- [35f9d472ba] Kevin Harwell -- codecs.conf.sample: Add sample and option descriptions for codec_opus
Category: Resources/res_ari
ASTERISK-26488: ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
Reported by: Matt Jordan
- [1a3e699316] Matt Jordan -- res/stasis: Add CLI commands for displaying/debugging ARI apps
Category: Resources/res_rtp_asterisk
ASTERISK-26418: res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
Reported by: Michael Walton
- [494bebeb6f] Michael Walton -- res_rtp_asterisk: Add ice_blacklist option
Bug
Category: Addons/chan_ooh323
ASTERISK-24400: ooh323 sends wrong hangup code
Reported by: Dmitry Melekhov
- [2849e596b8] Alexander Anikin -- chan_ooh323: Fixes to work right with Cisco devices
Category: Applications/app_dial
ASTERISK-26549: app_dial: When PickupChan() is used some channels may have incorrect device state
Reported by: Joshua Colp
- [b2a078efc9] Joshua Colp -- app_dial: Fix incorrect device state when channel is picked up.
Category: Applications/app_voicemail
ASTERISK-26503: app_voicemail: Asterisk crashes when MailboxExists is used
Reported by: Doug Lytle
- [ebc293e609] Joshua Colp -- app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS.
Category: Bridges/bridge_softmix
ASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt Jordan
- [62d60c1f58] Matt Jordan -- main/bridge_channel: Fix channel reference leak on video source
- [be254aa8e3] Matt Jordan -- main/bridge: Add some verbose logging for video source changes
- [ff96e37509] Matt Jordan -- bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source
Category: Channels/chan_dahdi
ASTERISK-26412: build: Prepare for gcc 6.2
Reported by: George Joseph
- [25895897f8] Kevin Harwell -- stasis_recording/stored: remove calls to deprecated readdir_r function.
Category: Channels/chan_pjsip
ASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard Mudgett
- [fb05acbb32] Richard Mudgett -- res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.
- [4fda9e7b0b] Richard Mudgett -- bundled pjproject: Fix DNS write to freed memory.
ASTERISK-26482: [patch] chan_pjsip: segfault on already disconnected session
Reported by: Alexei Gradinari
- [3e040685c7] Alexei Gradinari -- chan_pjsip: segfault on already disconnected session
ASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John Kiniston
- [8e3aee2036] snuffy -- Fix issue with CLI not returning to prompt after running "features show"
Category: Channels/chan_sip/General
ASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
Reported by: Michael Keuter
- [ac82b40bff] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"
ASTERISK-26476: chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
Reported by: Sergey Grachev
- [782dfa09a8] Grachev Sergey -- chan_sip: Incorrect display option Outbound reg. retry 403
ASTERISK-26457: [patch] force_rport,auto_comedia: No NAT detection triggered.
Reported by: Alexander Traud
- [ca1b197765] Alexander Traud -- chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia.
Category: Channels/chan_unistim
ASTERISK-26565: chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
Reported by: Jason
- [ba09e7a23f] Igor Goncharovskiy -- Fix closing rtp ports after call finished in chan_unistim.
Category: Codecs/codec_opus
ASTERISK-26520: codec_opus: Generated fmtp line has no content
Reported by: scgm11
- [d6783e8fd1] Mark Michelson -- res_format_attr_opus: Fix fmtp generation.
Category: Core/AstMM
ASTERISK-26526: [UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
Reported by: Badalian Vyacheslav
- [b2bf6c4c22] Corey Farrell -- vector: Prevent NULL argument to memcpy.
ASTERISK-26524: astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
Reported by: Corey Farrell
- [39ba7aa91f] Corey Farrell -- astobj2: Declare private variable data_size for AO2_DEBUG only.
Category: Core/Bridging
ASTERISK-26555: Multi-party Video: Fix some post Asterisk-11 regressions
Reported by: Matt Jordan
- [62d60c1f58] Matt Jordan -- main/bridge_channel: Fix channel reference leak on video source
- [be254aa8e3] Matt Jordan -- main/bridge: Add some verbose logging for video source changes
- [ff96e37509] Matt Jordan -- bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source
Category: Core/BuildSystem
ASTERISK-26608: Compile and link failures on OpenBSD
Reported by: snuffyASTERISK-26592: Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
Reported by: George Joseph
- [e2046bfaf3] gtjoseph -- cli: Fix ast_el_read_char to work with libedit >= 3.1
ASTERISK-22480: Embedded pjproject: build.mak contains hardcoded full path to version.mak
Reported by: Matt Jordan
- [162bb27cfb] gtjoseph -- pjproject_bundled: Remove usage of tar's --strip-components option
Category: Core/CodecInterface
ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
Reported by: Richard Mudgett
- [d89a1645fc] Richard Mudgett -- codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.
Category: Core/General
ASTERISK-26605: codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
Reported by: Richard Mudgett
- [d89a1645fc] Richard Mudgett -- codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.
ASTERISK-26509: A few non-critical deprecation warnings when building on Ubuntu 16.10
Reported by: Jonathan Harris
- [25895897f8] Kevin Harwell -- stasis_recording/stored: remove calls to deprecated readdir_r function.
Category: Core/ManagerInterface
ASTERISK-26556: manager: AMI version report same in Ast 13 & 14, despite Ast 14 syntax changes
Reported by: Michelle Dupuis
- [f1359c7dc8] Joshua Colp -- manager: Bump AMI version number.
ASTERISK-26537: AMI: NewConnectedLine event is not documented
Reported by: Etienne Lessard
- [d9f9691d31] Etienne Lessard -- manager: Add documentation for NewConnectedLine event.
Category: Core/RTP
ASTERISK-24274: [patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
Reported by: Frankie Chin
- [6445f21caa] Alexander Traud -- rtp_engine: Allow more than 32 dynamic payload types.
ASTERISK-26311: [patch] rtp_engine: Allow more than 32 dynamic payload types.
Reported by: Alexander Traud
- [6445f21caa] Alexander Traud -- rtp_engine: Allow more than 32 dynamic payload types.
Category: Core/Stasis
ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI calls
Reported by: Daniele Pallastrelli
- [cfede6a1fc] Joshua Colp -- res_stasis: Don't unsubscribe from a NULL bridge.
Category: Documentation
ASTERISK-26514: Super Awesome Company: Don't specify transport in pjsip.conf
Reported by: Rusty Newton
- [c2a2643c69] Rusty Newton -- SAC documentation: don't specify transports for endpoints and registrations
Category: Features
ASTERISK-26444: 'features show' command in CLI does not return prompt.
Reported by: John Kiniston
- [8e3aee2036] snuffy -- Fix issue with CLI not returning to prompt after running "features show"
Category: General
ASTERISK-26575: testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
Reported by: Joshua Colp
- [cc4a2c8c76] Joshua Colp -- res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp.
ASTERISK-25070: Fix FTBFS on Hurd
Reported by: Gabriele GiaconeASTERISK-26387: Asterisk segfaults shortly after starting even with no active calls.
Reported by: Harley Peters
- [79ac79ab03] Richard Mudgett -- bundled pjproject: Crashes while resolving DNS names.
ASTERISK-26513: tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
Reported by: Joshua Colp
- [b76afa5e4f] Corey Farrell -- Fix shutdown crash caused by modules being left open.
ASTERISK-26421: Segmentation Fault with ARI originate into mixing bridge with 43 clients
Reported by: Andrew Nagy
- [98378133c0] Mark Michelson -- ARI: Detect duplicate channel IDs
- [0abc71dfd6] Mark Michelson -- CDR: Alter destruction pattern for CDR chains.
ASTERISK-26480: [patch] CLI: core set debug: Auto-completes File not Module
Reported by: Alexander Traud
- [3938e3320f] Alexander Traud -- cli: Auto-complete File not Module for core set debug.
Category: Resources/res_agi
ASTERISK-26343: ASTERISK-25951 causes issues for callerid manipulation through agi
Reported by: Morten Tryfoss
- [ac6051c302] gtjoseph -- channel: Fix issues in hangup scenarios caused by frame deferral
- [0288fba2f0] Mark Michelson -- autoservice: Use frame deferral API
- [8d8323b142] Mark Michelson -- AGI: Only defer frames when in an interception routine.
- [4a8b1940b8] Mark Michelson -- Add API for channel frame deferral.
Category: Resources/res_ari_bridges
ASTERISK-26468: ari: Bridge events stop working after this sequence of ARI calls
Reported by: Daniele Pallastrelli
- [cfede6a1fc] Joshua Colp -- res_stasis: Don't unsubscribe from a NULL bridge.
Category: Resources/res_pjsip
ASTERISK-26516: pjsip: Memory corruption with possible memory leak.
Reported by: Richard Mudgett
- [fb05acbb32] Richard Mudgett -- res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak.
- [4fda9e7b0b] Richard Mudgett -- bundled pjproject: Fix DNS write to freed memory.
ASTERISK-26571: res_pjsip: Resolution incorrect when explicit IPv6 transport configured
Reported by: Joshua Colp
- [119e1fd6cf] Joshua Colp -- res_pjsip: Perform resolution when explicit IPv6 transport is used.
Category: Resources/res_pjsip_caller_id
ASTERISK-26307: res_pjsip_caller_id: Crash on outgoing change
Reported by: Bill Brigden
- [82ef2bb69d] Joshua Colp -- res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls.
Category: Resources/res_pjsip_outbound_publish
ASTERISK-26506: [patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
Reported by: Matt Krokosz
- [d3e0d5d40b] Matt Krokosz -- res_pjsip_outbound_publish: Fix crash when publishing device state.
Category: Resources/res_pjsip_sdp_rtp
ASTERISK-26541: res_pjsip_sdp_rtp: Restrict number of formats to maximum
Reported by: Joshua Colp
- [69196a8db4] Joshua Colp -- res_pjsip_sdp_rtp: Limit number of formats to defined maximum.
ASTERISK-26423: res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
Reported by: Andreas Wetzel
- [791d2319ce] Joshua Colp -- pjsip: Fix a few media bugs with reinvites and asymmetric payloads.
ASTERISK-26309: [patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
Reported by: Alexander Traud
- [110c18f413] Joshua Colp -- res_pjsip_sdp_rtp: Fix address family of explicit media_address.
- [47de0ee4f6] Joshua Colp -- pjsip: Support dual stack automatically.
Category: Third-Party/pjproject
ASTERISK-26510: pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
Reported by: George Joseph
- [162bb27cfb] gtjoseph -- pjproject_bundled: Remove usage of tar's --strip-components option
Category: pjproject/pjsip
ASTERISK-26344: Asterisk 13.11.0 + PJSIP crash
Reported by: Ian Gilmour
- [79ac79ab03] Richard Mudgett -- bundled pjproject: Crashes while resolving DNS names.
New Feature
Category: General
ASTERISK-26595: ARI: Add the ability to control the source of video in a multi-party mixing bridge
Reported by: Matt Jordan
- [62cbcb2e54] Matt Jordan -- res/ari/resource_bridges: Add the ability to manipulate the video source
ASTERISK-26470: ARI: Add an 'asterisk_id' field to outgoing events
Reported by: Matt Jordan
- [8bd6f695dc] Joshua Colp -- ari: Update model validator based on addition of asterisk_id.
- [ebcbc9ee34] Matt Jordan -- res/ari: Add the Asterisk EID field to outgoing events
Category: Resources/res_ari
ASTERISK-26492: ARI: Add ability to specify channel variables on websocket events
Reported by: Mark Michelson
- [eb5077fb26] Mark Michelson -- res_ari: Add support for channel variables in ARI events.
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_queue
ASTERISK-26462: [patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
Reported by: Leandro Dardini
- [ef8c54238c] Leandro Dardini -- app_queue: Added initialization for "context" parameter
Category: Channels/chan_multicast_rtp
ASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode
- [3135b15a54] Moises Silva -- chan_rtp: Set a sane default rtp engine for unicast.
Category: Core/BuildSystem
ASTERISK-26356: menuselect: invalid test for GTK2
Reported by: Tzafrir Cohen
- [47ba379e4c] Tzafrir Cohen -- menuselect: invalid test for GTK2
Category: Core/Jitterbuffer
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind
Reported by: Florian Loyau
- [ac82b40bff] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"
Category: Core/RTP
ASTERISK-25270: chan_sip: rtptimeout doesn't work at all when using JitterBuffers of any kind
Reported by: Florian Loyau
- [ac82b40bff] Kevin Harwell -- Revert "chan_sip: Fix lastrtprx always updated"
Category: Resources/res_rtp_multicast
ASTERISK-26439: chan_rtp: Crash when originating
Reported by: Kayode
- [3135b15a54] Moises Silva -- chan_rtp: Set a sane default rtp engine for unicast.
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 |
|---|
| 7af1aae57f | Mark Michelson | Bump ARI version to 2.0.0 |
| 7a665c5c6e | Mark Michelson | manager: update minor version |
| 0825528322 | gtjoseph | Revert "Revert "AGI: Only defer frames when in an interception routine."" |
| 6b2efc116f | gtjoseph | Revert "Revert "autoservice: Use frame deferral API"" |
| f6acb765a8 | gtjoseph | Revert "Revert "channel: Use frame deferral API for safe sleep."" |
| d6f9e2b54a | gtjoseph | file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type |
| fb9b867d7d | Matt Jordan | pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS |
| 46bedcbbad | Matt Jordan | apps/app_echo: Only relay a single video source change frame |
| 88111da235 | gtjoseph | Revert "Revert "Add API for channel frame deferral."" |
| d36695e0bb | Richard Mudgett | res_pjsip.c: Rework endpt_send_request() req_wrapper code. |
| add253cbd0 | Richard Mudgett | res_pjsip: Fix tdata leaks in off nominal paths. |
| 49dd637047 | Richard Mudgett | res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. |
| b640f18a44 | gtjoseph | Revert "Add API for channel frame deferral." |
| 7155020c96 | gtjoseph | Revert "AGI: Only defer frames when in an interception routine." |
| 8b9996a90c | gtjoseph | Revert "autoservice: Use frame deferral API" |
| 97679ee846 | gtjoseph | Revert "channel: Use frame deferral API for safe sleep." |
| dbcb958254 | gtjoseph | build: Fix default values for some SANITIZER options |
| 65a0b6bca2 | Mark Michelson | res_pjsip_session: Do not call session supplements when it's too late. |
| 1db9e7886c | Mark Michelson | channel: Use frame deferral API for safe sleep. |
| e57f994237 | Alexander Anikin | chan_ooh323: reset rrq count on gk registration |
| 0f627421b9 | Michael Kuron | automon: restore mixing of the both channels after recording stops |
| 59d25b2081 | Matt Jordan | res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems |
| d889fb375b | Matt Jordan | res_stasis: Set a video source mode on Stasis created bridges |
| e6b414a66a | Alexander Anikin | chan_ooh323: Fix infinite loop on read second part of H.225 packet |
| aa04c453b8 | gtjoseph | pjproject_bundled: Fix issue with libasteriskpj needing libresample |
| 23812d60b9 | gtjoseph | pjproject_bundled: Fix compile of pjsua so it handles audio |
| d84eaa46fa | gtjoseph | pjproject_bundled: Fix issue where "/version.mak" wasn't found |
| f2d406ced8 | gtjoseph | test_astobj2_thrash: Fix multithreaded issues |
| cb342298cd | Pascal Cadotte Michaud | typo: s/paranthesis/parenthesis/ in a comment |
| 0836b617e5 | gtjoseph | pjproject_bundled: Fixed various build issues |
| ba145944a0 | Mark Michelson | ARI: Add duplicate channel ID checking for channel creation. |
| 13aa241b9c | gtjoseph | utils.c: Fix ast_set_default_eid for multiple platforms |
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 |57142 ----------
asterisk-14.1.0-summary.html | 572
asterisk-14.1.0-summary.txt | 1360
b/CHANGES | 101
b/addons/ooh323c/src/ooCalls.c | 3
b/addons/ooh323c/src/ooGkClient.c | 1
b/addons/ooh323c/src/oochannels.c | 43
b/addons/ooh323c/src/ooq931.c | 5
b/apps/app_dial.c | 1
b/apps/app_echo.c | 3
b/apps/app_queue.c | 13
b/apps/app_voicemail.c | 2
b/bridges/bridge_builtin_features.c | 2
b/bridges/bridge_softmix.c | 28
b/channels/chan_pjsip.c | 237
b/channels/chan_rtp.c | 2
b/channels/chan_sip.c | 18
b/channels/chan_unistim.c | 11
b/configs/basic-pbx/pjsip.conf | 3
b/configs/samples/ari.conf.sample | 5
b/configs/samples/asterisk.conf.sample | 9
b/configs/samples/codecs.conf.sample | 54
b/configs/samples/pjsip.conf.sample | 11
b/configs/samples/rtp.conf.sample | 12
b/configure | 220
b/configure.ac | 12
b/contrib/ast-db-manage/config/versions/4468b4a91372_add_pjsip_asymmetric_rtp_codec.py | 31
b/contrib/scripts/sip_to_pjsip/sip_to_pjsip.py | 2
b/doc/appdocsxml.xslt | 20
b/include/asterisk.h | 9
b/include/asterisk/_private.h | 1
b/include/asterisk/autoconfig.h.in | 6
b/include/asterisk/bridge.h | 9
b/include/asterisk/channel.h | 91
b/include/asterisk/channel_internal.h | 2
b/include/asterisk/file.h | 28
b/include/asterisk/json.h | 12
b/include/asterisk/manager.h | 2
b/include/asterisk/module.h | 7
b/include/asterisk/options.h | 2
b/include/asterisk/res_pjsip.h | 2
b/include/asterisk/rtp_engine.h | 3
b/include/asterisk/stasis_app.h | 10
b/include/asterisk/stasis_bridges.h | 4
b/include/asterisk/stasis_channels.h | 1
b/include/asterisk/vector.h | 8
b/main/Makefile | 11
b/main/asterisk.c | 48
b/main/astobj2.c | 4
b/main/autoservice.c | 66
b/main/bridge.c | 34
b/main/bridge_channel.c | 3
b/main/cdr.c | 19
b/main/channel.c | 238
b/main/channel_internal_api.c | 28
b/main/cli.c | 14
b/main/codec_builtin.c | 16
b/main/features_config.c | 2
b/main/file.c | 137
b/main/format_cap.c | 2
b/main/json.c | 13
b/main/loader.c | 5
b/main/manager_bridges.c | 52
b/main/manager_channels.c | 11
b/main/netsock.c | 2
b/main/rtp_engine.c | 87
b/main/stasis_bridges.c | 29
b/main/stasis_channels.c | 5
b/main/utils.c | 244
b/menuselect/aclocal.m4 | 281
b/menuselect/configure | 197
b/menuselect/configure.ac | 9
b/res/ari/ari_model_validators.c | 481
b/res/ari/ari_model_validators.h | 67
b/res/ari/ari_websockets.c | 2
b/res/ari/config.c | 20
b/res/ari/resource_bridges.c | 66
b/res/ari/resource_bridges.h | 28
b/res/ari/resource_channels.c | 14
b/res/res_agi.c | 38
b/res/res_ari.c | 3
b/res/res_ari_bridges.c | 146
b/res/res_ari_channels.c | 3
b/res/res_format_attr_opus.c | 10
b/res/res_http_websocket.c | 19
b/res/res_pjsip.c | 137
b/res/res_pjsip/include/res_pjsip_private.h | 14
b/res/res_pjsip/pjsip_configuration.c | 1
b/res/res_pjsip/pjsip_message_ip_updater.c | 303
b/res/res_pjsip/pjsip_resolver.c | 73
b/res/res_pjsip_caller_id.c | 14
b/res/res_pjsip_outbound_authenticator_digest.c | 13
b/res/res_pjsip_outbound_publish.c | 1
b/res/res_pjsip_outbound_registration.c | 2
b/res/res_pjsip_pubsub.c | 20
b/res/res_pjsip_registrar_expire.c | 2
b/res/res_pjsip_sdp_rtp.c | 54
b/res/res_pjsip_session.c | 15
b/res/res_pjsip_t38.c | 13
b/res/res_rtp_asterisk.c | 107
b/res/res_stasis.c | 22
b/res/stasis/app.c | 105
b/res/stasis/app.h | 26
b/res/stasis/cli.c | 216
b/res/stasis/cli.h | 43
b/res/stasis_recording/stored.c | 217
b/rest-api/api-docs/applications.json | 2
b/rest-api/api-docs/asterisk.json | 2
b/rest-api/api-docs/bridges.json | 84
b/rest-api/api-docs/channels.json | 21
b/rest-api/api-docs/deviceStates.json | 2
b/rest-api/api-docs/endpoints.json | 2
b/rest-api/api-docs/events.json | 22
b/rest-api/api-docs/mailboxes.json | 2
b/rest-api/api-docs/playbacks.json | 2
b/rest-api/api-docs/recordings.json | 2
b/rest-api/api-docs/sounds.json | 2
b/rest-api/resources.json | 2
b/tests/test_astobj2_thrash.c | 11
b/tests/test_file.c | 197
b/tests/test_res_stasis.c | 6
b/third-party/pjproject/Makefile | 75
b/third-party/pjproject/Makefile.rules | 10
b/third-party/pjproject/apply_patches | 4
b/third-party/pjproject/configure.m4 | 5
b/third-party/pjproject/patches/0000-remove-third-party.patch | 142
b/third-party/pjproject/patches/0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch | 134
b/third-party/pjproject/patches/0006-r5473-svn-backport-Fix-pending-query.patch | 28
b/third-party/pjproject/patches/0006-r5475-svn-backport-Remove-DNS-cache-entry.patch | 30
contrib/realtime/mssql/mssql_cdr.sql | 58
contrib/realtime/mssql/mssql_config.sql | 1575
contrib/realtime/mssql/mssql_voicemail.sql | 54
contrib/realtime/mysql/mysql_cdr.sql | 40
contrib/realtime/mysql/mysql_config.sql | 968
contrib/realtime/mysql/mysql_voicemail.sql | 34
contrib/realtime/oracle/oracle_cdr.sql | 52
contrib/realtime/oracle/oracle_config.sql | 1569
contrib/realtime/oracle/oracle_voicemail.sql | 48
contrib/realtime/postgresql/postgresql_cdr.sql | 44
contrib/realtime/postgresql/postgresql_config.sql | 1046
contrib/realtime/postgresql/postgresql_voicemail.sql | 38
res/res_pjsip_multihomed.c | 225
144 files changed, 4903 insertions(+), 65609 deletions(-)