Changelog for asterisk-console-18.12.1-6.185.i586.rpm :

* Mon Apr 17 2023 Jan Engelhardt - Enable chan_mobile
* Thu May 19 2022 Michael Ströder - Updated pjproject to 2.12- Update to release 18.12.1
* Release 18.12.1: - [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing requests
* Release 18.12.0: + Security fixes - [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities - [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \\ - [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files + New Features - [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter but everyone else can - [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at prefix - [ASTERISK-29486] - Hint-like extension value lookup function without device state - [ASTERISK-29941] - chan_pjsip: Add ability to send flash events - [ASTERISK-29820] - cli: Add command to evaluate a function - [ASTERISK-29876] - app_queue: Add music on hold option + Bugs fixes - [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera available - [ASTERISK-29638] - res_pjsip_session: No video after early media - [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold - [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi restart - [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with missing secrets - [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled are always recompiled - [ASTERISK-30002] - app_meetme: Don\'t erroneously set global variables when channel is NULL - [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max number of groups - [ASTERISK-22246] - Asterisk\'s \"T\" flag is ignored when used with \"r\" or \"R\" flags. (documentation bug) - [ASTERISK-26582] - Asterisk seems to ignore the \"n\" parameter for \"disable console colorization\" - [ASTERISK-29843] - Session timers get removed on UPDATE - [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented - [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE - [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if early_media already enabled - [ASTERISK-29948] - iostream: Infinite TCP timeout writing data - [ASTERISK-29253] - Incorrect bridging on transfer - [ASTERISK-30006] - res_pjsip: UDP transport does not work when async_operations is greater than 1 - [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled - [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new keyword - [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database columns - [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP attributes - [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity - [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions - [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name - [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 - [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1) - [ASTERISK-29986] - build: Asterisk 18.11.0 doesn\'t compile when wget isn\'t available - [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn\'t - [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats printout - [ASTERISK-29939] - agi: Fix xmldoc bug with set music - [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments displayed incorreclty - [ASTERISK-29048] - chan_iax2: \"iax2 show registry\" shows host for perceived - [ASTERISK-29674] - Adjust for 64bit time_t - [ASTERISK-29961] - RLS: domain part of \'uri\' list attribute mismatch with SUBSCRIBE request - [ASTERISK-29928] - logging messages truncated when using MUSL runtime - [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file - [ASTERISK-29950] - SayNumber can handle \'01\' to \'07\', but not \'08\' or \'09\' + Improvements - [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring cadences - [ASTERISK-29940] - general: Add since tags to xmldocs - [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP) implementation - [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup - [ASTERISK-29954] - app_meetme: Emit warning if conference not found - [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk - [ASTERISK-29976] - Should Readme include information about install_prereq script? - [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries - [ASTERISK-29980] - build: External binary modules don\'t use https - [ASTERISK-25716] - Documentation: Document explanations and examples for possible values of DIALSTATUS - [ASTERISK-29967] - pbx_builtins: Add missing documentation
* Tue Apr 26 2022 Michael Ströder - Update to release 18.11.3
* [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled
* Fri Apr 15 2022 Michael Ströder - Update to release 18.11.2 with security fixes for
* AST-2022-001: res_stir_shaken: resource exhaustion with large files
* AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header
* AST-2022-003: func_odbc: Possible SQL Injection- remove unpackaged file
* Wed Mar 30 2022 Michael Ströder - Update to release 18.11.1
* [ASTERISK-29986] - build: Asterisk 18.11.0 doesn\'t compile when wget isn\'t available
* [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn\'t
* Thu Mar 24 2022 Michael Ströder - Updated to jansson-2.14- Update to release 18.11.0:
* Security bugs fixed: - [ASTERISK-29945] - pjproject: Security fixes for things
* New Features: - [ASTERISK-29853] - ami: Allow events to be globally disabled - [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
* Bugs fixed: - [ASTERISK-29924] - res_config_pgsql: omit \"unsupported column type \'text\'\" error - [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists - [ASTERISK-29904] - RLS: Batched Notifications stop working - [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown - [ASTERISK-29873] - [patch] Queue Realtime load - [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a call event. - [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) - [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs - [ASTERISK-29886] - Asterisk AMI sends not-valid XML
* Improvements: - [ASTERISK-29909] - app_queue: Add support for withdrawing a call - [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists - [ASTERISK-29353] - Qualify jansson 2.14 for asterisk - [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs - [ASTERISK-29896] - xmldocs: Add since tag - [ASTERISK-29861] - asterisk.h: add macro for curl user agent - [ASTERISK-29809] - curl, stir_shaken: refactor curl code - [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox - [ASTERISK-29925] - func_db: Warn about malformed key names - [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions - [ASTERISK-29866] - cli: add core dump information to core show settings - [ASTERISK-29898] - documentation: Add default attributes to documentation - [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility - [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits
* Sat Mar 05 2022 Michael Ströder - Update to release 18.10.1 also with many bug fixes and small improvements
* Security fixes: - AST-2022-004: pjproject: integer underflow on STUN message - AST-2022-005: pjproject: undefined behavior after freeing a dialog set - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
* New Features 18.10.0: - [ASTERISK-29808] cdr: allow disabling CDR by default - [ASTERISK-29830] ami: Add AMI event for Wink - [ASTERISK-29802] app_sf: Add full tech-agnostic SF support - [ASTERISK-29759] app_sendtext: Add ReceiveText application - [ASTERISK-29706] func_json: Add JSON parsing function
* Sun Feb 06 2022 Martin Hauke - Reenable build with support for DAHDI on supported platforms
* Sat Jan 08 2022 Michael Ströder - Update to release 18.9.0
* New Features - [ASTERISK-29720] - res_tonedetect: Add call progress tone detection - [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to Queue Members
* Bugs fixed - [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors. - [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen - [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning - [ASTERISK-29776] - stir/shaken: Requires GNU designator - [ASTERISK-29764] - chan_misdn: Fix for Doxygen - [ASTERISK-29773] - progdocs: doxyref.h outdated - [ASTERISK-29765] - xmldoc: Fix for Doxygen - [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes - [ASTERISK-29762] - channels: Fix for Doxygen - [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in same bridge - [ASTERISK-29754] - odbc: Fix for Doxygen - [ASTERISK-29753] - parking: Fix for Doxygen - [ASTERISK-29755] - frame: Fix for Doxygen - [ASTERISK-29756] - res_ari: Fix for Doxygen - [ASTERISK-29751] - channel: Fix for Doxygen - [ASTERISK-29750] - stasis: Fix for Doxygen - [ASTERISK-29752] - app: Fix for Doxygen - [ASTERISK-29749] - res_xmpp: Fix for Doxygen - [ASTERISK-29742] - addons: Fix for Doxygen. - [ASTERISK-29747] - res_pjsip: Fix for Doxygen - [ASTERISK-29737] - chan_iax2: Fix for Doxygen - [ASTERISK-29743] - bridges: Fix for Doxygen - [ASTERISK-29741] - tests: Fix for Doxygen - [ASTERISK-29740] - apps: Fix for Doxygen - [ASTERISK-29733] - progdocs: Avoid name with Doxygen \\file - [ASTERISK-29736] - bridge_channel: Fix for Doxygen - [ASTERISK-29735] - progdocs: Avoid multiple use of section labels - [ASTERISK-29734] - progdocs: Use Doxygen \\example correctly - [ASTERISK-29744] - app_morsecode: Fix deadlock - [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing - [ASTERISK-29705] - app_read: Fix custom terminator functionality regression - [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined. - [ASTERISK-29702] - sig_analog: Fix truncated buffer copy - [ASTERISK-28040] - pbx: \"dialplan reload\" is removing minus symbol from dynamic hints - [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup - [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions. - [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable dispatched. - [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized - [ASTERISK-29713] - GCC 11.2: two stringop-overread - [ASTERISK-29682] - Squash compiler issues generated by gcc 11 - [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile - [ASTERISK-27816] - func_talkdetect\'s logic is completely broken - [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request - [ASTERISK-26497] - make install downloads x86_32 variants of external modules on non Intel architectures
* Improvements - [ASTERISK-29777] - documentation: Standardize example syntax - [ASTERISK-29715] - app_voicemail: Refactor email generation functions - [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent - [ASTERISK-29714] - Spelling errors - [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at dial time - [ASTERISK-29662] - Add mix option to Playback application for say and filename
* Thu Sep 09 2021 Jan Engelhardt - Update to release 18.6.0
* AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
* app_reload: New Reload application
* app_waitforcond: New application
* app_dtmfstore: New application to store digits
* AST-2021-008 - chan_iax2: remote crash on unsupported media format Thu May 13 21:04:27 UTC 2021 - Diederik de Groot - Bug
* Category: Applications/app_queue ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not working Reported by: Michael
* [35302efe73] Sean Bright -- app_queue: Add alembic migration to add ringinuse to queue_members. ASTERISK-24631: Incorrect description of option \"context\" in queues.conf.sample Reported by: Etienne Lessard
* [31364fa4c8] Sean Bright -- queues.conf.sample: Correct \'context\' documentation. ASTERISK-26614: app_queue: updatecdr option in queues.conf does effectively nothing Reported by: Alexander Gonchiy
* [e27fa9eceb] Sean Bright -- app_queue.c: Remove dead \'updatecdr\' code. ASTERISK-27542: app_queue: When \"queue show\" CLI command is executed a crash occurs Reported by: Miguel Sanz
* [4393207751] 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
* [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus if status changes.
* Category: Bridges/bridge_simple ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590 Reported by: Ross Beer
* [88aec107df] George Joseph -- bridge_channel_write_frame: Check for NULL channel
* Category: Channels/chan_local ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing Reported by: Matthias Hensler
* [ed2f637b47] Joshua C. Colp -- core_unreal: Fix deadlock with T.38 control frames.
* Category: Core/BuildSystem ASTERISK-29348: menuselect doesn\'t return errors in many cases Reported by: George Joseph
* [f47c5cbdf9] Jaco Kroon -- menuselect: exit non-zero in case of failure on --enable|disable options.
* Category: Core/CodecInterface ASTERISK-29328: translate.c: possible buffer overflow when upsampling Reported by: Jean Aunis - Prescom
* [dec44306cf] Jean Aunis -- translate.c: Take sampling rate into account when checking codec\'s buffer size
* Category: Core/Stasis ASTERISK-29355: app_queue: Queue member status message sent even if status doesn\'t change Reported by: Roman Pertsev
* [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus if status changes.
* Category: Documentation ASTERISK-24434: Fix differing usage of assignment operators in modules.conf Reported by: Rusty Newton
* [be3153346b] Sean Bright -- modules.conf: Fix more differing usages of assignment operators. ASTERISK-24631: Incorrect description of option \"context\" in queues.conf.sample Reported by: Etienne Lessard
* [31364fa4c8] Sean Bright -- queues.conf.sample: Correct \'context\' documentation. ASTERISK-25358: dateformat not read from logger.conf by remote console Reported by: Igor Liferenko
* [a0009c807e] Mark Murawski -- logger: Console sessions will now respect logger.conf dateformat= option
* Category: Resources/General ASTERISK-29130: prometheus: Crash when scraping bridge Reported by: Francisco Correia
* [19eef2a6dc] George Joseph -- res_prometheus: Clone containers before iterating
* Category: Resources/res_pjsip ASTERISK-29354: res_pjsip: Allow partial reloading of transports Reported by: Joshua C. Colp
* [f213833514] Joshua C. Colp -- res_pjsip: Add support for partial transport reload.
* Category: Resources/res_pjsip_session ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused asterisk crash Reported by: sungtae kim
* [c78d0ce429] George Joseph -- res_pjsip_session: Make reschedule_reinvite check for NULL topologies
* Category: Resources/res_rtp_asterisk ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation Reported by: Kevin Harwell
* [17c86dcfaa] Kevin Harwell -- res_rtp_asterisk: Fix standard deviation calculation ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated Reported by: N A
* [b0d828f14a] 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
* [2e7fc84398] Joshua C. Colp -- res_rtp_asterisk: Force resync on SSRC change.- Improvement
* Category: Core/General ASTERISK-29339: loader: Let\'s output warnings for deprecated modules! Reported by: Joshua C. Colp
* [a9a9864478] 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
* [6aac148d59] Joshua C. Colp -- menuselect: Add ability to set deprecated and removed versions.
* [60fb559ccc] 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
* [60800b038a] Joshua C. Colp -- xml: Embed module information into core XML documentation.
* Category: Documentation ASTERISK-29336: documentation: Fix inconsistent support levels Reported by: Joshua C. Colp
* [be3e469f98] Joshua C. Colp -- documentation: Fix non-matching module support levels. ASTERISK-29335: xml: Embed module information into core XML documentation. Reported by: Joshua C. Colp
* [60800b038a] Joshua C. Colp -- xml: Embed module information into core XML documentation.
* Fri Mar 26 2021 Michael Ströder - update to 18.3.0
* app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and MixMonitorMute when the channel monitoring is started, stopped and muted (or unmuted) respectively.
* chan_iax2: You can now specify a default \"auth\" method in the [general] section of iax.conf
* chan_pjsip, app_transfer: Added TRANSFERSTATUSPROTOCOL variable. performing a REFER.
* Introduce an ARGC variable for func_odbc functions, along with a minargs per-function configuration option.
* SRTP replay protection has been added to res_srtp and a new configuration option \"srtpreplayprotection\" has been added to the rtp.conf config file.
* Sun Mar 14 2021 Jan Engelhardt - Update to release 18.2.2
* AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
* Thu Feb 18 2021 Michael Ströder - Update to 18.2.1 with security fixes:
* AST-2021-001: Remote crash in res_pjsip_diversion
* AST-2021-002: Remote crash possible when negotiating T.38
* AST-2021-003: Remote attacker could prematurely tear down SRTP calls
* AST-2021-004: An unsuspecting user could crash Asterisk with multiple
* AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver
* Sat Feb 13 2021 Jan Engelhardt - Cut build recipe parts for platforms older than SLE/Leap 15
* Sat Feb 13 2021 Asterisk Team - update to 18.2.0:
* Security - [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
* Bug fixes - [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event - [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy - [ASTERISK-29155] - app_queue: Deadlock between queues container and individual queues - [ASTERISK-29161] - Incorrect setup of recall channels - [ASTERISK-29168] - Asterisk crashes during call transfer - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable - [ASTERISK-27902] - chan_pjsip isn\'t updating hangupcause on 4XX responses - [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses - [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped - [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if registration can\'t be send - [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the Transfer result - [ASTERISK-29210] - res_pjsip: Crash when examining transport - [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. - [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. - [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. - [ASTERISK-29209] - Debug messages printed by scope trace might be missing newlines - [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels spuriously - [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend - [ASTERISK-29188] - null media causing the Asterisk crash - [ASTERISK-29173] - Media cache URL requests allow infinite redirects - [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without entries - [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS response - [ASTERISK-29191] - tel: URI in Diversion header causes crash - [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable - [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription - [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description - [ASTERISK-29191] - tel: URI in Diversion header causes crash - [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
* Improvements - [ASTERISK-29118] - VoiceMail() should have an option to play greetings as Early Media - [ASTERISK-28549] - Two repeated 183 - [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other newer linux versions - [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in /tmp - [ASTERISK-28549] - Two repeated 183
* Tue Dec 22 2020 Torrey Searle - Update for 18.1.1:
* Security bugs fixed: - [AST-2020-001] - res_pjsip: Return dialog locked and referenced - [AST-2020-002] - res_pjsip: Stop sending INVITEs after challenge limit.
* Thu Nov 19 2020 Michael Ströder - update to 17.9.0:
* Security bugs fixed: - [ASTERISK-29057] - pjsip: Crash on call rejection during high load
* Improvements: - [ASTERISK-29055] - Create a Bridge with video_single mode - [ASTERISK-29056] - Increase reg_server column size for ps_contacts table realtime
* many more bug fixes
* Fri Nov 06 2020 Michael Ströder - update to 17.8.1 with security fixes:
* AST-2020-001: Remote crash in res_pjsip_session
* AST-2020-002: Outbound INVITE loop on challenge with different nonce.
* Fri Oct 23 2020 Hans-Peter Jansen - Asterisk 17.8.0
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand)
* ASTERISK-29043 - app_queue: Leave empty sometimes not recorded as abandoned (Reported by Kfir Itzhak)
* ASTERISK-29042 - res_parking: Parker UUID is no longer copied (Reported by Misha Vodsedalek)
* ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported by Joseph Ades)
* 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)
* ASTERISK-29040 - res_speech: Assertion on format (Reported by Nickolay V. Shmyrev)
* ASTERISK-29001 - chan_pjsip does not process or forward 181 responses (Reported by Torrey Searle)
* ASTERISK-29034 - Lastpause of realtime members is reseting (Reported by Evandro César Arruda)
* 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)
* ASTERISK-29033 - res_pjsip_session: Aggressively terminates session on failed re-INVITE (Reported by Joshua C. Colp)
* ASTERISK-28974 - res_rtp_asterisk: T.140 messages have appended RTP string to each message block. (Reported by Thomas Johnson)
* Mon Oct 19 2020 Hans-Peter Jansen - Add dahdi build conditional dahdi-linux is bitrotten, and TW kernel is moving too fast to catch up- Use proper gmime dependency- Add full asterisk include folder
* Thu Sep 03 2020 Michael Ströder - Update to release 17.7.0
* [ASTERISK-29042] - res_parking: Parker UUID is no longer copied
* [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously doing an ExtensionState on a pattern match hint that ends up adding an extension
* [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload
* [ASTERISK-29021] - Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
* [ASTERISK-28927] - Asterisk crash in music on hold
* [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address)
* [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured contacts is not correct
* [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
* [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on reading rule from realtime
* [ASTERISK-28975] - res_http_websocket: Text payload data doesn\'t necessary include trailing zero
* Fri Jul 17 2020 Diederik de Groot - Update to release 17.6.0
* AMI: You can now specify an optional \'Content-Type\' as an argument for the Asterisk SendText manager action.
* res_pjsip: Added a new PJSIP system setting called disable_rport.
* res_sorcery_memory_cache: The SorceryMemoryCacheExpireObject AMI action and CLI command allow expiring of a specific object within the sorcery memory cache.
* res_ari_channels: When creating a channel in ARI using the create call you can now specify dialplan variables to be set as part of the same operation.
* res_pjsip_logger: The PJSIP packet logger now has the following CLI commands:
* Sat Jun 06 2020 Jan Engelhardt - Update to release 17.4.0
* ARI: Application event filtering is now supported. An application can now specify an \"allowed\" and/or \"disallowed\" list(s) of event types.
* AttendedTransfer: A new application, this will queue up attended transfer to the given extension.
* BlindTransfer: A new application, this will redirect all channels currently bridged to the caller channel to the specified destination.
* ConfBridge: Add \"average_all\", \"highest_all\", and \"lowest_all\" values for the remb_behavior option. These values operate on a bridge level instead of a per-source level.
* Dial: Add RINGTIME and RINGTIME_MS variables containing respectively seconds and milliseconds between creation of the dialing channel and receiving the first RINGING signal.
* Dial: Add PROGRESSTIME and PROGRESSTIME_MS variables analogous to the above with respect to the PROGRESS signal. Shorter of these two times should be equivalent to the PDD (Post Dial Delay) value.
* Dial: Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get millisecond resolution versions of DIALEDTIME and ANSWEREDTIME.
* Thu Mar 12 2020 Hans-Peter Jansen - Update to new upstream release 16.8.0 + Bugs fixed in this release:
* ASTERISK-28766 - PJSIP blind transfer not completed after using Proceeding() (Reported by lvl)
* ASTERISK-28685 - check_expr2: linking (when hardening) and cross-compiling troubles (Reported by Sebastian Kemper)
* ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and seqno handling (Reported by Joshua C. Colp)
* ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in the \"variables\" field (Reported by Jean Aunis - Prescom)
* ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After Hold (Reported by Ross Beer)
* ASTERISK-28697 - res_pjsip: Named ACL does not update on reload if changed (Reported by Timothy Vanderaerden)
* ASTERISK-28746 - res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set (Reported by George Joseph)
* ASTERISK-28716 - ICE: pjnath shouldn\'t wait for ICE to complete before allowing sending (Reported by Benjamin Keith Ford)
* ASTERISK-28738 - Incorrect state machine used when MOH_PASSTHRU is used (Reported by Torrey Searle)
* ASTERISK-28742 - res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup (Reported by Kevin Harwell)
* ASTERISK-28735 - Realtime MoH Unknown format \'\' -- defaulting to SLIN (Reported by Ross Beer)
* ASTERISK-28730 - res_pjsip_session: Fix out of order session refreshes (Reported by Joshua C. Colp)
* ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are depleted, should return 503 (Reported by Walter Doekes)
* ASTERISK-28719 - Cannot remove defaultrule from queue using realtime queues (Reported by EDV O-TON)
* ASTERISK-28713 - res_stasis_playback: Error building JSON (Reported by Sébastien Duthil)
* ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
* ASTERISK-26082 - res_pjsip_messaging: MessageSend Content-Type can\'t be changed (Reported by Alex)
* ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer)
* ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn)
* ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the error when sending (Reported by Dmitriy Serov)
* ASTERISK-28686 - chan_sip strictrtp=yes fails when media source is changed: no audio (Reported by Walter Doekes)
* ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes Asterisk To Drop Calls (Reported by Paul Brooks)
* ASTERISK-26955 - pjsip: SIP Packets with Via \"received=\" Containing IPv6 Address Delimited by \"[]\" Rejected (Reported by Peter Sokolov) + Improvements made in this release:
* ASTERISK-28750 - TLS/SSL Key too small error (Reported by Martin Zeh)
* ASTERISK-28733 - stream: Add support for adding/removing streams during SFU/calls (Reported by Joshua C. Colp)
* ASTERISK-24798 - Documentation - Clarify That Format Is Set By File Name Extension In MixMonitor (Reported by xrobau)
* ASTERISK-28726 - install_prereq script uses the interactive mode when installing aptitude (Reported by Sylvain Afchain)
* Wed Feb 05 2020 Hans-Peter Jansen - Update to new upstream release 16.8.0 + New Features made in this release:
* ASTERISK-17491 - CURLOPT() needs a \"followlocation\" parameter / \"maxredirs\" doesn\'t do anything (Reported by candrews)
* ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add ability to match on source port (Reported by Sean Bright) + Bugs fixed in this release:
* ASTERISK-28679 - stasis application is destroyed after its creation (Reported by Francois Blackburn)
* ASTERISK-28423 - ARI causes STASIS Deadlock (Reported by Ross Beer)
* ASTERISK-28714 - REGRESSION: Feature subscription_persistence_recreate (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
* ASTERISK-28677 - CDR billsec is always 0 for transferred calls (Reported by Maciej Michno)
* ASTERISK-28702 - chan_dahdi: holding a channel via flash to dialtone times out after 0:16:40 (Reported by Andrew Siplas)
* ASTERISK-28706 - silk 24hHz doesn\'t show up in \'core show translation\' output (Reported by Sean Bright)
* ASTERISK-24484 - Update documentation for statsd module - usage requirements unclear (Reported by Dan Jenkins)
* ASTERISK-28695 - core: minmemfree watermark uses free RAM, not available RAM (Reported by Kevin Flyn)
* ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a whitespace appears empty in the dialplan (Reported by Frank Matano)
* ASTERISK-23739 - [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used (Reported by Stas Kobzar)
* ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen)
* ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun)
* ASTERISK-21794 - CLI command \'realtime update2\' syntax failure when using according to usage help (Reported by Cedric BASSAGET)
* ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp)
* ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk)
* ASTERISK-28682 - app_record: Lack of `beep` audio file causes application to return error and hangup (Reported by Corey Farrell)
* ASTERISK-28507 - Wiki docs missing for MessageWaiting (Reported by David M. Lee)
* ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence does not preserve XML version number (Reported by Bryan Nelson)
* ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios with concurrent command pri show span X (Reported by Dirk Wendland)
* ASTERISK-28633 - stasis bridge topic leak (Reported by Joeran Vinzens)
* ASTERISK-28492 - pjsip reload not reloading wizard endpoint/pickup_group endpoint/call_group (Reported by Jean-Denis Girard)
* ASTERISK-28562 - SIP WSS message not processed until next frame arrives (Reported by Robert Sutton)
* ASTERISK-27243 - contrib: valgrind.supp doesn\'t suppress what it\'s supposed to due to invalid syntax (Reported by Richard Kenner)
* ASTERISK-28497 - func_odbc: truncating Unicode string on readsql (Reported by Boris P. Korzun)
* ASTERISK-28647 - chan_sip: RTP frames not transmitted after emitting a COLP (Reported by Jean Aunis - Prescom)
* ASTERISK-28667 - Asterisk ignores parsing of config files if a Byte order mark is present (Reported by Robin Leffmann)
* ASTERISK-28664 - \"trustrpid\" is misspelled in sip_to_pjsip.py (Reported by Pascal Cadotte Michaud)
* ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don\'t build on 17.0.0 (Reported by George Joseph)
* ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes non-existent media stream if codecs create additional streams and offer does not have them (Reported by nappsoft)
* ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation with config option (Reported by Kevin Harwell)
* ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes fails to deactivate CDR. (Reported by Frederic LE FOLL)
* ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function documentation (Reported by Pascal Cadotte Michaud)
* ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
* ASTERISK-28651 - chan_sip logs errors on tx to non-existent TCP connections (Reported by Jaco Kroon)
* ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER 200 Response Contact (Reported by Ross Beer)
* ASTERISK-28625 - Playback of local files impacted by large media cache (Reported by Kevin Reeves) + Improvements made in this release:
* ASTERISK-28710 - Should be able to disable the /httpstatus URI in the built-in HTTP server (Reported by Sean Bright)
* ASTERISK-28638 - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported by cmaj)
* ASTERISK-28673 - GET FULL VARIABLE documentation clarification (Reported by Jonathan Harris)
* ASTERISK-28658 - app_confbridge: Add support for setting maximum sample rate (Reported by Joshua C. Colp)