Release Summary asterisk-13.17.0 Date: 2017-07-12 ---------------------------------------------------------------------- Table of Contents 1. Summary 2. Contributors 3. Closed Issues 4. Open Issues 5. Other Changes 6. 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-13.16.0. ---------------------------------------------------------------------- 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 17 Sean Bright 4 Alexei Gradinari 12 George Joseph 4 Joshua Colp 10 Joshua Colp 3 Kevin Harwell 9 Alexei Gradinari 3 Louis Jocelyn Paquet 5 Richard Mudgett 3 Tzafrir Cohen 5 Kevin Harwell 3 George Joseph 2 Torrey Searle 2 Guido Falsi 2 Guido Falsi 2 Alexander Traud 2 Alexander Traud 2 Michael Walton 1 Jan Friesse 2 Torrey Searle 1 Florian Floimair 1 Rusty Newton 1 Ivan Poddubny 1 Matthew Fredrickson 1 Matthew Fredrickson 1 Jacek Konieczny 1 Yasin CANER 1 Tim Morgan 1 David M. Lee 1 Etienne Allovon 1 Robert Mordec 1 alex 1 JA,rgen H 1 Kinsey Moore 1 Rodrigo Ramirez Norambuena 1 John Harris 1 Frederic LE FOLL 1 Javier Riveros 1 Corey Farrell 1 Sean Bright 1 Robert Mordec 1 Ross Beer 1 Chris Howard 1 mdu113 1 Andrew Nowrot 1 'alex' 1 Lorne Gaetz 1 Ben Langfeld 1 John Fawcett 1 Corey Farrell 1 Frankie Chin 1 Zach R 1 Matthias Binder 1 Christopher van de Sande 1 Stefan EngstrAP:m 1 Antoine Pitrou 1 Alex 1 Etienne Lessard 1 Ryan Smith 1 Michael Maier 1 OpenBSD ports 1 Marek Cervenka 1 Ronald Raikes 1 Ove Aursand 1 Richard Mudgett 1 Frederic LE FOLL 1 wushumasters 1 Tony Mountifield 1 JA,rgen H 1 Michel R. Vaillancourt 1 David Brillert 1 Yasin CANER ---------------------------------------------------------------------- Closed Issues [Back to Top] This is a list of all issues from the issue tracker that were closed by changes that went into this release. Bug Category: Addons/format_mp3 ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded Reported by: Tzafrir Cohen * [97b003f5e2] Sean Bright -- format_mp3: Re-work menuselect/build issues * [72213c98e3] Sean Bright -- format_mp3: Don't try to build format_mp3 if we don't have sources Category: Applications/app_confbridge ASTERISK-27012: app_confbridge: ConfBridge sometimes does not play user name recording while leaving Reported by: Robert Mordec * [f1b32de2c5] Robert Mordec -- app_confbridge: Race between removing and playing name recording while leaving Category: Applications/app_meetme ASTERISK-27025: channel / meetme: Fix missing parentheses Reported by: Joshua Colp * [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses. Category: Applications/app_queue ASTERISK-25665: Duplicate logging in queue log for EXITEMPTY events Reported by: Ove Aursand * [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty ASTERISK-27065: call hangup after leaving app_queue Reported by: Marek Cervenka * [2c43ca0ac5] Ivan Poddubny -- app_queue: Fix returning to dialplan when a queue is empty ASTERISK-26399: app_queue: Agent not called when caller is parked Reported by: wushumasters * [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in call when not. ASTERISK-26400: app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime Reported by: Etienne Lessard * [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in call when not. ASTERISK-26715: app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel Reported by: David Brillert * [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in call when not. ASTERISK-26975: app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call Reported by: Lorne Gaetz * [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in call when not. Category: Applications/app_voicemail/IMAP ASTERISK-24052: app_voicemail reloads result in leaked IMAP sockets. Reported by: Louis Jocelyn Paquet * [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection control * [3b6c327c51] Alexei Gradinari -- app_voicemail: IMAP logout on reload/unload * [08be5e01e8] Alexei Gradinari -- app_voicemail: IMAP logout on MWI unsubscribe Category: Bridges/bridge_simple ASTERISK-26973: bridge: Crash when freeing frame and snooping Reported by: Michel R. Vaillancourt * [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks Category: Channels/chan_pjsip ASTERISK-27039: chan_pjsip: Device state is idle when channel from endpoint is in early media Reported by: Joshua Colp * [1f10c6b3b0] Joshua Colp -- chan_pjsip: Update device state when in early media. ASTERISK-26996: chan_pjsip: Flipping between codecs Reported by: Michael Maier * [996a4791ff] Joshua Colp -- pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. ASTERISK-26281: chan_pjsip would send INVITE to 'Unreachable' endpoints Reported by: Jacek Konieczny * [746c2c5745] Joshua Colp -- res_pjsip: Add support for returning only reachable contacts and use it. Category: Channels/chan_sip/General ASTERISK-27106: [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. Reported by: Alexander Traud * [39d2ebbf56] Alexander Traud -- chan_sip: Only when different, add TCP|TLS in autodomain (SIP Domain Support). * [9f4b3b966e] Alexander Traud -- chan_sip: Fix a typo for tlsbindaddr in autodomain (SIP Domain Support). ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable Reported by: Stefan EngstrAP:m * [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX Category: Channels/chan_sip/SRTP ASTERISK-25101: DTLS configuration can not be specified in the general section - documentation Reported by: Ben Langfeld * [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted Category: Codecs/General ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec Reported by: Frankie Chin * [70e5887906] Sean Bright -- format: Reintroduce smoother flags Category: Core/Bridging ASTERISK-27075: bridge: stuck channel(s) after failed attended transfer Reported by: Kevin Harwell * [67664fbf95] Kevin Harwell -- bridge: stuck channel(s) after failed attended transfer ASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge Reported by: Torrey Searle * [e414833f6e] Joshua Colp -- bridge: Add a deferred queue. Category: Core/Channels ASTERISK-27100: channel: ast_waitfordigit_full fails to clear flag in an error branch. Reported by: Corey Farrell * [73520e9f58] Corey Farrell -- channel: Clear channel flag in error branch. ASTERISK-27074: core_local: local channel data not being properly unref'ed and unlocked Reported by: Kevin Harwell * [1f9913f272] Kevin Harwell -- core_local: local channel data not being properly unref'ed and unlocked ASTERISK-26923: bridging: T.38 request is lost when channels are added to bridge Reported by: Torrey Searle * [e414833f6e] Joshua Colp -- bridge: Add a deferred queue. ASTERISK-27025: channel / meetme: Fix missing parentheses Reported by: Joshua Colp * [dc05183f4b] Joshua Colp -- channel / app_meetme: Fix parentheses. Category: Core/General ASTERISK-26789: Audit manipulation of channel flags without locks Reported by: Joshua Colp * [1618203964] Joshua Colp -- asterisk: Audit locking of channel when manipulating flags. Category: Core/PBX ASTERISK-27041: Core/PBX: [patch] Deadlock between dialplan execution and application unregistration Reported by: Frederic LE FOLL * [dc307af7f2] Frederic LE FOLL -- Core/PBX: Deadlock between dialplan execution and application unregistration. Category: Core/RTP ASTERISK-26978: rtp: Crash in ast_rtp_codecs_payload_code() Reported by: Ross Beer * [eb48e99bd4] George Joseph -- bridge_native_rtp: Keep rtp instance refs on bridge_channel ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec Reported by: Frankie Chin * [70e5887906] Sean Bright -- format: Reintroduce smoother flags Category: Core/Sorcery ASTERISK-27057: Seg Fault in ast_sorcery_object_get_id at sorcery.c Reported by: Ryan Smith * [c2eea791e4] George Joseph -- res_pjsip_pubsub: Fix reference to released endpoint Category: Documentation ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP argument Reported by: Rusty Newton * [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation Category: General ASTERISK-27108: Crash using 'data get' CLI command Reported by: Sean Bright * [6258de458b] Sean Bright -- core: Fix segfault when invoking 'data get' CLI command ASTERISK-27060: Comment typo format_g729.c Reported by: Matthew Fredrickson * [0a40073750] Matthew Fredrickson -- formats/format_g729: Fix typo in comment Category: PBX/pbx_realtime ASTERISK-19291: Background in realtime Reported by: Andrew Nowrot * [283cc59af7] Sean Bright -- pbx_builtin: Properly handle hangup during Background Category: Resources/res_agi ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP argument Reported by: Rusty Newton * [3eb7fbba72] Sean Bright -- res_agi: Clarify 'RECORD FILE' documentation ASTERISK-22432: Async AGI crashes Asterisk when issuing "set variable" command without args Reported by: Antoine Pitrou * [f306e451f6] Sean Bright -- res_agi: Prevent crash when SET VARIABLE called without arguments ASTERISK-25662: Malformed AGI 520 Usage response Reported by: Tony Mountifield * [a007e438c3] Sean Bright -- res_agi: Fix malformed AGI usage response Category: Resources/res_ari ASTERISK-27026: res_ari: Crash when no ari.conf configuration file exists Reported by: Ronald Raikes * [7901b9853e] George Joseph -- res_ari: Add "module loaded" check to ari stubs Category: Resources/res_ari_recordings ASTERISK-27021: GET /recordings/stored returns 500 Internal Server Error Reported by: Tim Morgan * [cf6cf59646] Sean Bright -- stasis_recording: Correct ast_asprintf error checking Category: Resources/res_format_attr_h264 ASTERISK-27008: res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space Reported by: John Harris * [700ef6861a] Sean Bright -- res_format_attr_h26x: Trim blanks in fmtp attributes Category: Resources/res_parking ASTERISK-26399: app_queue: Agent not called when caller is parked Reported by: wushumasters * [6bfcb1acc7] Joshua Colp -- app_queue: Fix members showing as being in call when not. Category: Resources/res_pjsip ASTERISK-27090: PJSIP: Deadlock using TCP transport Reported by: Richard Mudgett * [0d64cbde57] Richard Mudgett -- pjsip_distributor.c: Fix deadlock with TCP type transports. Category: Resources/res_pjsip/Bundling ASTERISK-27052: Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network Reported by: alex * [0bde568669] George Joseph -- pjproject_bundled: Use the asterisk github mirror for download Category: Resources/res_pjsip_refer ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer Reported by: Kevin Harwell * [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer Category: Resources/res_pjsip_session ASTERISK-27024: nat/external_media settings ignored in 14.4.1 Reported by: Christopher van de Sande * [2dee95cc7a] Florian Floimair -- res_pjsip_session: Correct inverted test in session_outgoing_nat_hook ASTERISK-27053: res_pjsip_refer/session: Calls dropped during transfer Reported by: Kevin Harwell * [6cdf3191d3] Kevin Harwell -- res_pjsip_refer/session: Calls dropped during transfer ASTERISK-26964: res_pjsip_session: Wrong From on reinvite when request and To URI differ Reported by: Yasin CANER * [36628cc9c4] Yasin CANER -- res_pjsip_session : fixed wrong From Header number On Re-invite Category: Resources/res_pjsip_transport_websocket ASTERISK-27046: res_pjsip_transport_websocket: segfault in get_write_timeout Reported by: JA,rgen H * [e16a669c70] JA,rgen H -- res_pjsip_transport_websocket: Add NULL check in get_write_timeout Category: Resources/res_rtp_asterisk ASTERISK-27022: res_rtp_asterisk: Incorrect SSRC change for RTCP component Reported by: Michael Walton * [7dafe82751] George Joseph -- res_rtp_asterisk: Fix ssrc change for rtcp srtp ASTERISK-24858: [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec Reported by: Frankie Chin * [70e5887906] Sean Bright -- format: Reintroduce smoother flags ASTERISK-25101: DTLS configuration can not be specified in the general section - documentation Reported by: Ben Langfeld * [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 Reported by: Javier Riveros * [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm ASTERISK-26982: chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable Reported by: Stefan EngstrAP:m * [4479038073] Sean Bright -- chan_sip: Better ICE handling for RTCP-MUX Category: Resources/res_srtp ASTERISK-25294: srtp's crypto_get_random deprecated Reported by: Tzafrir Cohen * [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2 ASTERISK-25101: DTLS configuration can not be specified in the general section - documentation Reported by: Ben Langfeld * [971a401ce9] Sean Bright -- sip.conf.sample: Clarify where DTLS settings are permitted ASTERISK-26979: res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110 Reported by: Javier Riveros * [e91efef2bb] Kevin Harwell -- res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm Category: Resources/res_stasis_snoop ASTERISK-26973: bridge: Crash when freeing frame and snooping Reported by: Michel R. Vaillancourt * [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks Category: pjproject/pjsip ASTERISK-26333: Problems with Blind Transfer, PJSIP (Aastra 6869i) Reported by: Matthias Binder * [6af2dd34af] Alexei Gradinari -- res_pjsip: New endpoint option "refer_blind_progress" Improvement Category: Core/BuildSystem ASTERISK-27043: Core/BuildSystem: Add defines to fix build with LibreSSL Reported by: Guido Falsi * [6a64f65fe6] Guido Falsi -- BuildSystem: Add patches to allow building with recent LibreSSL Category: Core/Channels ASTERISK-26419: audiohooks: Remove redundant codec translations when using audiohooks Reported by: Michael Walton * [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks Category: Core/General ASTERISK-26419: audiohooks: Remove redundant codec translations when using audiohooks Reported by: Michael Walton * [adfb28882b] Kevin Harwell -- channel: ast_write frame wrongly freed after call to audiohooks Category: Core/Portability ASTERISK-27042: Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file Reported by: Guido Falsi * [44cee2f4a1] Guido Falsi -- BuildSystem: Fix build on FreeBSD due to missing crypt.h Category: Resources/res_agi ASTERISK-26124: res_agi: Set audio format for EAGI audio stream Reported by: John Fawcett * [90237dca11] Sean Bright -- res_agi: Allow configuration of audio format of EAGI pipe Category: Resources/res_pjsip_mwi ASTERISK-26230: [patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup Reported by: Alexei Gradinari * [0f6a9617eb] Alexei Gradinari -- res_pjsip_mwi: update unsolicited MWI subscriptions on updating contact * [59c9bbe696] Alexei Gradinari -- res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled Category: Resources/res_rtp_asterisk ASTERISK-26976: libsrtp-2.x.x support Reported by: Alex * [5e9cd1f20d] Sean Bright -- res_srtp: Add support for libsrtp2 ---------------------------------------------------------------------- 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: Bridges/bridge_simple ASTERISK-26469: Infinite loop after a dual Redirect Reported by: Etienne Allovon * [b07b216235] Joshua Colp -- manager: Clear the flag on the other channel. Category: Channels/chan_pjsip ASTERISK-27095: chan_pjsip: When connected_line_method is set to invite, we're not trying UPDATE Reported by: George Joseph * [6bd7c0f37c] George Joseph -- chan_pjsip: Fix ability to send UPDATE on COLP Category: Core/Bridging ASTERISK-27016: Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times. Reported by: Chris Howard * [4910a3bf40] Joshua Colp -- channel: Fix reference counting in ast_channel_suppress. Category: General ASTERISK-27088: res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation Reported by: Joshua Colp * [0426b1d88a] Joshua Colp -- res_rtp_asterisk: Fix issues with ICE renegotiation. Category: Resources/res_corosync ASTERISK-25370: res_corosync segfaults at startup with corosync version > 2.x Reported by: mdu113 * [005a4afa6b] Jan Friesse -- res_corosync: Change thread stack size Category: Resources/res_pjsip_dialog_info_body_generator ASTERISK-26919: res_pjsip_dialog_info_body_generator: Ringing&&InUse behavior difference between chan_sip and res_pjsip Reported by: Zach R * [a6e4899612] Alexei Gradinari -- res_pjsip: New endpoint option "notify_early_inuse_ringing" Category: Resources/res_pjsip_mwi ASTERISK-27051: res_pjsip_mwi: unsolicited MWI has to be unsubscribed on deleting the endpoint's last contact Reported by: Alexei Gradinari * [8e749c8f51] Alexei Gradinari -- res_pjsip_mwi: unsubscribe unsolicited MWI on deleting endpoint last contact Category: Resources/res_stasis ASTERISK-27059: res_stasis: Stolen channel references are leaking Reported by: George Joseph * [edfdb4dff5] George Joseph -- res_stasis: Plug reference leak on stolen channels Category: Third-Party/pjproject ASTERISK-27097: pjproject_bundled: We don't pass options needed for cross-compile to pjproject configure Reported by: George Joseph * [bbe68f139d] George Joseph -- pjproject_bundled: Allow passing configure options to bundled Improvement Category: Applications/app_voicemail/IMAP ASTERISK-27068: app_voicemail: Add global option "imap_poll_logout" to specify post-polling disconnect Reported by: Alexei Gradinari * [8f356192d1] Alexei Gradinari -- app_voicemail: IMAP connection control Category: Channels/chan_pjsip ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode Reported by: Torrey Searle * [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode Category: Resources/res_pjsip ASTERISK-27066: res_pjsip: Add DTMF INFO Failback mode Reported by: Torrey Searle * [9fbc34d2bd] Torrey Searle -- res_pjsip: Add DTMF INFO Failback mode ---------------------------------------------------------------------- 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 | |------------+------------------+----------------------------------------| | 0c00ee754b | George Joseph | Update for 13.17.0-rc1 | |------------+------------------+----------------------------------------| | 379fe65831 | George Joseph | Fix alembic branches | |------------+------------------+----------------------------------------| | 905d18e8bf | Richard Mudgett | pjsip_distributor.c: Fix | | | | unidentified_requests hash functions. | |------------+------------------+----------------------------------------| | 1f59d08924 | Torrey Searle | res/res_pjsip_t38: fix incorrect | | | | increment of media_count | |------------+------------------+----------------------------------------| | 764d04fa87 | Richard Mudgett | res_pjsip_mwi.c: Eliminate RAII_VAR in | | | | contact delete observer | |------------+------------------+----------------------------------------| | cecf6540dc | Rodrigo RamArez | cdr: fix mistake spelling of a word | | | Norambuena | for Unanswered. | |------------+------------------+----------------------------------------| | b9a4ab8c8c | Richard Mudgett | chan_pjsip: Fix PJSIP_MEDIA_OFFER | | | | dialplan function read. | |------------+------------------+----------------------------------------| | f1a209d5ac | Richard Mudgett | app_voicemail.c: Fix compile error | | | | when IMAP enabled. | |------------+------------------+----------------------------------------| | 68de35a6a0 | David M. Lee | CFLAGS for BIND8 support | |------------+------------------+----------------------------------------| | da3312457e | Sean Bright | codecs.conf.sample: Fix max_bandwidth | | | | speling error | |------------+------------------+----------------------------------------| | 590ffcaf0b | Sean Bright | eventfd: Disable during cross | | | | compilation | |------------+------------------+----------------------------------------| | 5520b6c201 | Alexei Gradinari | CHANGES: correct version for a new | | | | option 'refer_blind_progress' | |------------+------------------+----------------------------------------| | c093bf8072 | Sean Bright | res_rtp_multicast: Use consistent | | | | timestamps when possible | |------------+------------------+----------------------------------------| | c10341646d | George Joseph | test_json: Fix test names with | | | | reserved words | |------------+------------------+----------------------------------------| | 65898c3af8 | George Joseph | unittests: Add a unit test that causes | | | | a SEGV and... | +------------------------------------------------------------------------+ ---------------------------------------------------------------------- Diffstat Results [Back to Top] This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility. asterisk-13.16.0-summary.html | 405 --- asterisk-13.16.0-summary.txt | 952 --------- b/.version | 2 b/CHANGES | 54 b/ChangeLog | 1045 +++++++++- b/Makefile | 3 b/addons/Makefile | 10 b/apps/app_chanspy.c | 16 b/apps/app_confbridge.c | 79 b/apps/app_dial.c | 6 b/apps/app_disa.c | 10 b/apps/app_dumpchan.c | 4 b/apps/app_externalivr.c | 6 b/apps/app_meetme.c | 2 b/apps/app_queue.c | 109 - b/apps/app_voicemail.c | 80 b/asterisk-13.17.0-rc1-summary.html | 311 ++ b/asterisk-13.17.0-rc1-summary.txt | 832 +++++++ b/autoconf/ast_ext_lib.m4 | 36 b/bridges/bridge_native_rtp.c | 677 +++++- b/bridges/bridge_simple.c | 32 b/channels/chan_pjsip.c | 68 b/channels/chan_sip.c | 8 b/channels/pjsip/dialplan_functions.c | 37 b/configs/samples/cdr.conf.sample | 2 b/configs/samples/codecs.conf.sample | 6 b/configs/samples/pjsip.conf.sample | 20 b/configs/samples/sip.conf.sample | 3 b/configs/samples/voicemail.conf.sample | 3 b/configure | 434 +++- b/configure.ac | 100 b/contrib/ast-db-manage/config/versions/164abbd708c_add_auto_info_to_endpoint_dtmf_mode.py | 58 b/contrib/ast-db-manage/config/versions/86bb1efa278d_add_ps_endpoints_refer_blind_progress.py | 30 b/contrib/ast-db-manage/config/versions/d7983954dd96_add_ps_endpoints_notify_early_inuse_.py | 30 b/contrib/realtime/mssql/mssql_config.sql | 46 b/contrib/realtime/mysql/mysql_config.sql | 18 b/contrib/realtime/oracle/oracle_config.sql | 46 b/contrib/realtime/postgresql/postgresql_config.sql | 22 b/formats/format_g729.c | 2 b/include/asterisk/ari.h | 10 b/include/asterisk/autoconfig.h.in | 3 b/include/asterisk/bridge_channel.h | 2 b/include/asterisk/bridge_channel_internal.h | 11 b/include/asterisk/bridge_technology.h | 3 b/include/asterisk/channel.h | 25 b/include/asterisk/codec.h | 3 b/include/asterisk/core_local.h | 37 b/include/asterisk/format.h | 11 b/include/asterisk/res_pjsip.h | 74 b/include/asterisk/res_pjsip_presence_xml.h | 3 b/include/asterisk/res_pjsip_session.h | 11 b/include/asterisk/rtp_engine.h | 9 b/include/asterisk/smoother.h | 1 b/include/asterisk/test.h | 8 b/main/autoservice.c | 2 b/main/bridge.c | 10 b/main/bridge_after.c | 2 b/main/bridge_channel.c | 38 b/main/channel.c | 90 b/main/codec_builtin.c | 19 b/main/core_local.c | 54 b/main/crypt.c | 2 b/main/data.c | 4 b/main/file.c | 20 b/main/format.c | 8 b/main/libasteriskssl.c | 4 b/main/manager.c | 8 b/main/pbx.c | 4 b/main/pbx_app.c | 7 b/main/pbx_builtins.c | 8 b/main/tcptls.c | 4 b/main/test.c | 4 b/makeopts.in | 2 b/res/res_agi.c | 73 b/res/res_ari_applications.c | 4 b/res/res_ari_asterisk.c | 4 b/res/res_ari_bridges.c | 4 b/res/res_ari_channels.c | 4 b/res/res_ari_device_states.c | 4 b/res/res_ari_endpoints.c | 4 b/res/res_ari_events.c | 33 b/res/res_ari_mailboxes.c | 4 b/res/res_ari_playbacks.c | 4 b/res/res_ari_recordings.c | 4 b/res/res_ari_sounds.c | 4 b/res/res_corosync.c | 29 b/res/res_format_attr_h263.c | 2 b/res/res_format_attr_h264.c | 2 b/res/res_musiconhold.c | 4 b/res/res_pjsip.c | 31 b/res/res_pjsip/location.c | 53 b/res/res_pjsip/pjsip_configuration.c | 9 b/res/res_pjsip/pjsip_distributor.c | 242 +- b/res/res_pjsip/presence_xml.c | 9 b/res/res_pjsip_dialog_info_body_generator.c | 10 b/res/res_pjsip_mwi.c | 87 b/res/res_pjsip_pidf_body_generator.c | 2 b/res/res_pjsip_pidf_eyebeam_body_supplement.c | 2 b/res/res_pjsip_pubsub.c | 8 b/res/res_pjsip_refer.c | 28 b/res/res_pjsip_sdp_rtp.c | 38 b/res/res_pjsip_session.c | 37 b/res/res_pjsip_session.exports.in | 1 b/res/res_pjsip_t38.c | 2 b/res/res_pjsip_transport_websocket.c | 4 b/res/res_pjsip_xpidf_body_generator.c | 2 b/res/res_rtp_asterisk.c | 41 b/res/res_rtp_multicast.c | 139 + b/res/res_srtp.c | 15 b/res/res_stasis.c | 20 b/res/srtp/srtp_compat.h | 29 b/res/stasis_recording/stored.c | 4 b/rest-api-templates/res_ari_resource.c.mustache | 35 b/tests/test_bridging.c | 292 ++ b/tests/test_json.c | 16 b/tests/test_pbx.c | 22 b/third-party/configure.m4 | 5 b/third-party/pjproject/Makefile | 2 b/third-party/pjproject/Makefile.rules | 7 b/third-party/pjproject/configure.m4 | 24 b/third-party/pjproject/patches/0070-Set-PJSIP_INV_SUPPORT_UPDATE-correctly-in-pjsip_inv_.patch | 16 121 files changed, 5477 insertions(+), 2043 deletions(-)