diff --git a/.version b/.version
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+++ b/.version
@@ -1 +1 @@
-certified-20.7-cert1-rc2
+certified-20.7-cert1
diff --git a/CHANGES.md b/CHANGES.md
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@@ -1 +1 @@
-ChangeLogs/ChangeLog-certified-20.7-cert1-rc2.md
\ No newline at end of file
+ChangeLogs/ChangeLog-certified-20.7-cert1.md
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@@ -0,0 +1,19263 @@
+
+## Change Log for Release asterisk-certified-20.7-cert1
+
+### Links:
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-20.7-cert1.md)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/certified-18.9-cert10...certified-20.7-cert1)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-20.7-cert1.tar.gz)
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/certified-asterisk)
+
+### Summary:
+
+- Commits: 1097
+- Commit Authors: 114
+- Issues Resolved: 891
+- Security Advisories Resolved: 1
+ - [GHSA-hxj9-xwr8-w8pq](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq): Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation
+
+### User Notes:
+
+- #### app_voicemail_odbc: Allow audio to be kept on disk
+ This commit adds a new voicemail.conf option
+ 'odbc_audio_on_disk' which when set causes the ODBC variant of
+ app_voicemail_odbc to leave the message and greeting audio files
+ on disk and only store the message metadata in the database.
+ Much more information can be found in the voicemail.conf.sample
+ file.
+
+- #### tcptls/iostream: Add support for setting SNI on client TLS connections
+ Secure websocket client connections now send SNI in
+ the TLS client hello.
+
+- #### app_dial: Add dial time for progress/ringing.
+ The timeout argument to Dial now allows
+ specifying the maximum amount of time to dial if
+ early media is not received.
+
+- #### app_voicemail: Allow preventing mark messages as urgent.
+ The leaveurgent mailbox option can now be used to
+ control whether callers may leave messages marked as 'Urgent'.
+
+- #### Stir/Shaken Refactor
+ Asterisk's stir-shaken feature has been refactored to
+ correct interoperability, RFC compliance, and performance issues.
+ See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
+ information.
+
+- #### Upgrade bundled pjproject to 2.14.
+ Bundled pjproject has been upgraded to 2.14. For more
+ information on what all is included in this change, check out the
+ pjproject Github page: https://github.com/pjsip/pjproject/releases
+
+- #### app_speech_utils.c: Allow partial speech results.
+ The SpeechBackground dialplan application now supports a 'p'
+ option that will return partial results from speech engines that
+ provide them when a timeout occurs.
+
+- #### app_chanspy: Add 'D' option for dual-channel audio
+ The ChanSpy application now accepts the 'D' option which
+ will interleave the spied audio within the outgoing frames. The
+ purpose of this is to allow the audio to be read as a Dual channel
+ stream with separate incoming and outgoing audio. Setting both the
+ 'o' option and the 'D' option and results in the 'D' option being
+ ignored.
+
+- #### chan_dahdi: Allow MWI to be manually toggled on channels.
+ The 'dahdi set mwi' now allows MWI on channels
+ to be manually toggled if needed for troubleshooting.
+ Resolves: #440
+
+- #### app_dial: Add option "j" to preserve initial stream topology of caller
+ The option "j" is now available for the Dial application which
+ uses the initial stream topology of the caller to create the outgoing
+ channels.
+
+- #### logger: Add channel-based filtering.
+ The console log can now be filtered by
+ channels or groups of channels, using the
+ logger filter CLI commands.
+
+- #### chan_pjsip: Add PJSIPHangup dialplan app and manager action
+ A new dialplan app PJSIPHangup and AMI action allows you
+ to hang up an unanswered incoming PJSIP call with a specific SIP
+ response code in the 400 -> 699 range.
+
+- #### app_voicemail: Add AMI event for mailbox PIN changes.
+ The VoicemailPasswordChange event is
+ now emitted whenever a mailbox password is updated,
+ containing the mailbox information and the new
+ password.
+ Resolves: #398
+
+- #### res_speech: allow speech to translate input channel
+ res_speech now supports translation of an input channel
+ to a format supported by the speech provider, provided a translation
+ path is available between the source format and provider capabilites.
+
+- #### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..
+ With this update, the PJSIP realm lengths have been extended
+ to support up to 255 characters.
+
+- #### res_stasis: signal when new command is queued
+ Call setup times should be significantly improved
+ when using ARI.
+
+- #### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
+ You no longer need to select DEBUG_THREADS to use
+ DETECT_DEADLOCKS. This removes a significant amount of overhead
+ if you just want to detect possible deadlocks vs needing full
+ lock tracing.
+
+- #### file.c: Add ability to search custom dir for sounds
+ A new option "sounds_search_custom_dir" has been added to
+ asterisk.conf that allows asterisk to search
+ AST_DATA_DIR/sounds/custom for sounds files before searching the
+ standard AST_DATA_DIR/sounds/<lang> directory.
+
+- #### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
+ The "Build Options" entry in the "core show settings"
+ CLI command has been renamed to "ABI related Build Options" and
+ a new entry named "All Build Options" has been added that shows
+ both breaking and non-breaking options.
+
+- #### chan_rtp: Implement RTP glue for UnicastRTP channels
+ The dial string option 'g' was added to the UnicastRTP channel
+ which enables RTP glue and therefore native RTP bridges with those
+ channels.
+
+- #### app_queue: periodic announcement configurable start time.
+ Introduce a new queue configuration option called
+ 'periodic-announce-startdelay' which will vary the normal (historic)
+ behavior of starting the periodic announcement cycle at
+ periodic-announce-frequency seconds after entering the queue to start
+ the periodic announcement cycle at period-announce-startdelay seconds
+ after joining the queue. The default behavior if this config option is
+ not set remains unchanged.
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+- #### variables: Add additional variable dialplan functions.
+ Four new dialplan functions have been added.
+ GLOBAL_DELETE and DELETE have been added which allows
+ the deletion of global and channel variables.
+ GLOBAL_EXISTS and VARIABLE_EXISTS have been added
+ which checks whether a global or channel variable has
+ been set.
+
+- #### sig_analog: Add Called Subscriber Held capability.
+ Called Subscriber Held is now supported for analog
+ FXS channels, using the calledsubscriberheld option. This allows
+ a station user to go on hook when receiving an incoming call
+ and resume from another phone on the same line by going on hook,
+ without disconnecting the call.
+
+- #### res_pjsip_header_funcs: Make prefix argument optional.
+ The prefix argument to PJSIP_HEADERS is now
+ optional. If not specified, all header names will be
+ returned.
+
+- #### core/ari/pjsip: Add refer mechanism
+ There is a new ARI endpoint `/endpoints/refer` for referring
+ an endpoint to some URI or endpoint.
+
+- #### chan_dahdi: Allow autoreoriginating after hangup.
+ The autoreoriginate setting now allows for kewlstart FXS
+ channels to automatically reoriginate and provide dial tone to the
+ user again after all calls on the line have cleared. This saves users
+ from having to manually hang up and pick up the receiver again before
+ making another call.
+
+- #### sig_analog: Allow three-way flash to time out to silence.
+ The threewaysilenthold option now allows the three-way
+ dial tone to time out to silence, rather than continuing forever.
+
+- #### res_pjsip: Enable TLS v1.3 if present.
+ res_pjsip now allows TLS v1.3 to be enabled if supported by
+ the underlying PJSIP library. The bundled version of PJSIP supports
+ TLS v1.3.
+
+- #### app_queue: Add support for applying caller priority change immediately.
+ The 'queue priority caller' CLI command and
+ 'QueueChangePriorityCaller' AMI action now have an 'immediate'
+ argument which allows the caller priority change to be reflected
+ immediately, causing the position of a caller to move within the
+ queue depending on the priorities of the other callers.
+
+- #### Adds manager actions to allow move/remove/forward individual messages in a par..
+ The following manager actions have been added
+ VoicemailBoxSummary - Generate message list for a given mailbox
+ VoicemailRemove - Remove a message from a mailbox folder
+ VoicemailMove - Move a message from one folder to another within a mailbox
+ VoicemailForward - Copy a message from one folder in one mailbox
+ to another folder in another or the same mailbox.
+
+- #### app_voicemail: add CLI commands for message manipulation
+ The following CLI commands have been added to app_voicemail
+ voicemail show mailbox <mailbox> <context>
+ Show contents of mailbox <mailbox>@<context>
+ voicemail remove <mailbox> <context> <from_folder> <messageid>
+ Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
+ voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
+ Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
+ voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
+ Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
+ mailbox <mailbox>@<context> <to_folder>
+
+- #### sig_analog: Allow immediate fake ring to be suppressed.
+ The immediatering option can now be set to no to suppress
+ the fake audible ringback provided when immediate=yes on FXS channels.
+
+- #### AMI: Add parking position parameter to Park action
+ New ParkingSpace parameter has been added to AMI action Park.
+
+- #### res_musiconhold: Add option to loop last file.
+ The loop_last option in musiconhold.conf now
+ allows the last file in the directory to be looped once reached.
+
+- #### AMI: Add CoreShowChannelMap action.
+ New AMI action CoreShowChannelMap has been added.
+
+- #### sig_analog: Add fuller Caller ID support.
+ Additional Caller ID properties are now supported on
+ incoming calls to FXS stations, namely the
+ redirecting reason and call qualifier.
+
+- #### res_stasis.c: Add new type 'sdp_label' for bridge creation.
+ When creating a bridge using the ARI the 'type' argument now
+ accepts a new value 'sdp_label' which will configure the bridge to add
+ labels for each stream in the SDP with the corresponding channel id.
+
+- #### app_queue: Preserve reason for realtime queues
+ Make paused reason in realtime queues persist an
+ Asterisk restart. This was fixed for non-realtime
+ queues in ASTERISK_25732.
+
+- #### cel: add local optimization begin event
+ The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
+ by itself or in conert with the existing
+ AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
+
+- #### chan_dahdi: Add dialmode option for FXS lines.
+ A "dialmode" option has been added which allows
+ specifying, on a per-channel basis, what methods of
+ subscriber dialing (pulse and/or tone) are permitted.
+ Additionally, this can be changed on a channel
+ at any point during a call using the CHANNEL
+ function.
+
+
+### Upgrade Notes:
+
+- #### pbx_variables.c: Prevent SEGV due to stack overflow.
+ The maximum amount of dialplan recursion
+ using variable substitution (such as by using EVAL_EXTEN)
+ is capped at 15.
+
+- #### Stir/Shaken Refactor
+ The stir-shaken refactor is a breaking change but since
+ it's not working now we don't think it matters. The
+ stir_shaken.conf file has changed significantly which means that
+ existing ones WILL need to be changed. The stir_shaken.conf.sample
+ file in configs/samples/ has quite a bit more information. This is
+ also an ABI breaking change since some of the existing objects
+ needed to be changed or removed, and new ones added. Additionally,
+ if res_stir_shaken is enabled in menuselect, you'll need to either
+ have the development package for libjwt v1.15.3 installed or use
+ the --with-libjwt-bundled option with ./configure.
+
+- #### app.c: Allow ampersands in playback lists to be escaped.
+ Ampersands in URLs passed to the `Playback()`,
+ `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
+ `Queue()` applications as filename arguments can now be escaped by
+ single quoting the filename. Additionally, this is also possible when
+ using the `CONFBRIDGE` dialplan function, or configuring various
+ features in `confbridge.conf` and `queues.conf`.
+
+- #### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
+ The dtls_rekey will be disabled if webrtc support is
+ requested on an endpoint. A warning will also be emitted.
+
+- #### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..
+ As part of this update, the maximum allowable length
+ for PJSIP endpoints and relevant resources has been increased from
+ 40 to 255 characters. To take advantage of this enhancement, it is
+ recommended to run the necessary procedures (e.g., Alembic) to
+ update your schemas.
+
+- #### app_queue: Preserve reason for realtime queues
+ Add a new column to the queue_member table:
+ reason_paused VARCHAR(80) so the reason can be preserved.
+
+- #### cel: add local optimization begin event
+ The existing AST_CEL_LOCAL_OPTIMIZE can continue
+ to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
+ can be ignored if desired.
+
+
+### Commit Authors:
+
+- Alex2grad: (1)
+- Alexander Greiner-Baer: (1)
+- Alexander Traud: (67)
+- Alexandre Fournier: (1)
+- Alexei Gradinari: (12)
+- Andre Barbosa: (3)
+- Andrew Siplas: (1)
+- Bastian Triller: (2)
+- Ben Ford: (27)
+- Bernd Zobl: (2)
+- Boris P. Korzun: (10)
+- Brad Smith: (4)
+- Carlos Oliva: (1)
+- Christof Efkemann: (1)
+- Cmaj: (3)
+- Dan Cropp: (2)
+- Dennis Buteyn: (1)
+- Dovid Bender: (1)
+- Dustin Marquess: (1)
+- Eduardo: (1)
+- Evandro C��Sar Arruda: (1)
+- Evgenios_Greek: (1)
+- Fabrice Fontaine: (2)
+- Florentin Mayer: (1)
+- Frederic LE FOLL: (1)
+- Frederic Van Espen: (1)
+- George Joseph: (139)
+- Gitea: (1)
+- Guido Falsi: (1)
+- Henning Westerholt: (3)
+- Holger Hans Peter Freyther: (9)
+- Hugh McMaster: (1)
+- Igor Goncharovsky: (5)
+- Ivan Poddubny: (1)
+- Ivan Poddubnyi: (5)
+- Jaco Kroon: (22)
+- Jason D. McCormick: (1)
+- Jasper Hafkenscheid: (1)
+- Jasper Van Der Neut: (1)
+- Jean Aunis: (4)
+- Jeremy Lain��: (1)
+- Jiajian Zhou: (1)
+- Joe Searle: (1)
+- Jose Lopes: (1)
+- Joseph Nadiv: (3)
+- Josh Soref: (25)
+- Joshua C. Colp: (85)
+- Joshua Colp: (1)
+- Kevin Harwell: (28)
+- Kfir Itzhak: (2)
+- Luke Escude: (1)
+- Lvl: (2)
+- Marcel Wagner: (3)
+- Mark Murawski: (4)
+- Mark Petersen: (10)
+- Martin Nystroem: (1)
+- Matthew Fredrickson: (4)
+- Matthew Kern: (1)
+- Maximilian Fridrich: (13)
+- Michael Cargile: (1)
+- Michael Kuron: (2)
+- Michael Neuhauser: (2)
+- Michal Hajek: (1)
+- Micha�� G��Rny: (5)
+- Miguel Angel Nubla: (1)
+- Mike Bradeen: (44)
+- MikeNaso: (1)
+- Moritz Fain: (1)
+- Nathan Bruning: (2)
+- Naveen Albert: (268)
+- Nick French: (4)
+- Nickolay Shmyrev: (1)
+- Nico Kooijman: (1)
+- Niklas Larsson: (1)
+- Olaf Titz: (1)
+- Patrick Verzele: (1)
+- Peter Fern: (1)
+- PeterHolik: (2)
+- Philip Prindeville: (14)
+- Phoneben: (1)
+- Pirmin Walthert: (1)
+- Richard Mudgett: (2)
+- Rijnhard Hessel: (1)
+- Roadkill: (1)
+- Robert Cripps: (1)
+- Rodrigo Ram��Rez Norambuena: (1)
+- Romryz: (1)
+- Salah Ahmed: (1)
+- Sam Banks: (1)
+- Samuel Olaechea: (1)
+- Sarah Autumn: (1)
+- Sean Bright: (155)
+- Sebastian Jennen: (1)
+- Sebastien Duthil: (4)
+- Sergey V. Lobanov: (1)
+- Shaaah: (1)
+- Shloime Rosenblum: (3)
+- Shyju Kanaprath: (1)
+- Spiridonov Dmitry: (1)
+- Stanislav Abramenkov: (2)
+- Stanislav: (1)
+- Steve Davies: (1)
+- Sungtae Kim: (10)
+- The_Blode: (1)
+- Thomas Guebels: (1)
+- Tinet-Mucw: (1)
+- Torrey Searle: (6)
+- Trevor Peirce: (2)
+- Under: (1)
+- Vitezslav Novy: (1)
+- Walter Doekes: (1)
+- Yury Kirsanov: (1)
+- Zhengsh: (3)
+- Zhou_jiajian: (1)
+
+## Issue and Commit Detail:
+
+### Closed Issues:
+
+ - !GHSA-hxj9-xwr8-w8pq: Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation
+ - 35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
+ - 37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms
+ - 39: [Bug]: Remove .gitreview from repository.
+ - 43: [Bug]: Link to trademark policy is no longer correct
+ - 45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is insufficient/ineffective
+ - 46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
+ - 48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
+ - 52: [improvement]: Add local optimization begin cel event
+ - 55: [bug]: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
+ - 64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
+ - 65: [bug]: heap overflow by default at startup
+ - 66: [improvement]: Fix preserve reason of pause when Asterisk is restared for realtime queues
+ - 71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource
+ - 73: [new-feature]: pjsip: Allow topology/session refreshes in early media state
+ - 84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
+ - 87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
+ - 89: [improvement]: indications: logging changes
+ - 91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP labels
+ - 94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
+ - 96: [bug]: make install-logrotate causes logrotate to fail on service restart
+ - 98: [new-feature]: callerid: Allow timezone to be specified at runtime
+ - 100: [bug]: sig_analog: hidecallerid setting is broken
+ - 102: [bug]: Strange warning - 'T' option is not compatible with remote console mode and has no effect.
+ - 104: [improvement]: Add AMI action to get a list of connected channels
+ - 108: [new-feature]: fair handling of calls in multi-queue scenarios
+ - 110: [improvement]: utils - add lock timing information with DEBUG_THREADS
+ - 116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
+ - 118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
+ - 120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
+ - 122: [new-feature]: res_musiconhold: Add looplast option
+ - 129: [bug]: res_speech_aeap: Crash due to NULL format on setup
+ - 133: [bug]: unlock channel after moh state access
+ - 136: [bug]: Makefile downloader does not follow redirects.
+ - 145: [bug]: ABI issue with pjproject and pjsip_inv_session
+ - 155: [bug]: GCC 13 is catching a few new trivial issues
+ - 158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
+ - 170: [improvement]: app_voicemail - add CLI commands to manipulate messages
+ - 174: [bug]: app_voicemail imap compile errors
+ - 179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
+ - 181: [improvement]: app_voicemail - add manager actions to display and manipulate messages
+ - 193: [bug]: third-party/apply-patches doesn't sort the patch file list before applying
+ - 200: [bug]: Regression: In app.h an enum is used before its declaration.
+ - 202: [improvement]: app_queue: Add support for immediately applying queue caller priority change
+ - 205: [new-feature]: sig_analog: Allow flash to time out to silent hold
+ - 224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
+ - 226: [improvement]: Apply contact_user to incoming calls
+ - 230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
+ - 233: [bug]: Deadlock with MixMonitorMute AMI action
+ - 240: [new-feature]: sig_analog: Add Called Subscriber Held capability
+ - 242: [new-feature]: logger: Allow filtering logs in CLI by channel
+ - 248: [bug]: core_local: Local channels cannot have slashes in the destination
+ - 253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs
+ - 255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered"
+ - 260: [bug]: maxptime must be changed to multiples of 20
+ - 263: [bug]: download_externals doesn't always handle versions correctly
+ - 265: [bug]: app_macro isn't locking around channel datastore access
+ - 267: [bug]: ari: refer with display_name key in request body leads to crash
+ - 274: [bug]: Syntax Error in SQL Code
+ - 275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement'
+ - 277: [bug]: pbx.c: Compiler error with gcc 12.2
+ - 281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits
+ - 286: [improvement]: chan_iax2: Improve authentication debugging
+ - 289: [new-feature]: Add support for deleting channel and global variables
+ - 294: [improvement]: chan_dahdi: Improve call pickup documentation
+ - 298: [improvement]: chan_rtp: Implement RTP glue
+ - 301: [bug]: Number of ICE TURN threads continually growing
+ - 303: [bug]: SpeechBackground never exits
+ - 308: [bug]: chan_console: Deadlock when hanging up console channels
+ - 315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/<lang>
+ - 316: [bug]: Privilege Escalation in Astrisk's Group permissions.
+ - 319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel
+ - 321: [bug]: Performance suffers unnecessarily when debugging deadlocks
+ - 325: [bug]: hangup after beep to avoid waiting for timeout
+ - 330: [improvement]: Add cel user event helper function
+ - 337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases
+ - 341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
+ - 345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality
+ - 349: [improvement]: Add libjwt to third-party
+ - 351: [improvement]: Refactor res_stir_shaken to use libjwt
+ - 352: [bug]: Update qualify_timeout documentation to include DNS note
+ - 354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
+ - 356: [new-feature]: app_directory: Add ADSI support.
+ - 360: [improvement]: Update documentation for CHANGES/UPGRADE files
+ - 362: [improvement]: Speed up ARI command processing
+ - 379: [bug]: Orphaned taskprocessors cause shutdown delays
+ - 384: [bug]: Unnecessary re-INVITE after answer
+ - 388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats
+ - 396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided
+ - 398: [new-feature]: app_voicemail: Add AMI event for password change
+ - 406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
+ - 409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
+ - 423: [improvement]: func_lock: Add missing see-also refs
+ - 425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample
+ - 428: [bug]: cli: Output is truncated from "config show help"
+ - 430: [bug]: Fix broken links
+ - 440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
+ - 442: [bug]: func_channel: Some channel options are not settable
+ - 445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases
+ - 458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
+ - 462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller
+ - 465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations
+ - 480: [improvement]: pbx_variables.c: Prevent infinite recursion and stack overflow with variable expansion
+ - 482: [improvement]: manager.c: Improve clarity of "manager show connected" output
+ - 492: [improvement]: res_calendar_icalendar: Print icalendar error if available on parsing failure
+ - 500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
+ - 503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't
+ - 505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr()
+ - 509: [bug]: res_pjsip: Crash when looking up transport state in use
+ - 513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG
+ - 520: [improvement]: menuselect: Use more specific error message.
+ - 527: [bug]: app_voicemail_odbc no longer working after removal of macrocontext.
+ - 529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed assertion bad magic number 0x0 for object" after ~30 calls
+ - 530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
+ - 533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of channel frame queue
+ - 539: [bug]: Existence of logger.xml causes linking failure
+ - 551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
+ - 560: [bug]: EndIf() causes next priority to be skipped
+ - 565: [bug]: Application Read() returns immediately
+ - 569: [improvement]: Add option to interleave input and output frames on spied channel
+ - 572: [improvement]: Copy partial speech results when Asterisk is ready to move on but the speech backend is not
+ - 582: [improvement]: Reduce unneeded logging during startup and shutdown
+ - 586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in older gcc versions
+ - 588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not received
+ - 592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault when a fax session is active
+ - 595: [improvement]: dsp.c: Fix and improve confusing warning message.
+ - 597: [bug]: wrong MOS calculation
+ - 601: [new-feature]: translate.c: implement new direct comp table mode (PR #585)
+ - 619: [new-feature]: app_voicemail: Allow preventing callers from marking messages as urgent
+ - 629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
+ - 634: [bug]: make install doesn't create the stir_shaken cache directory
+ - 636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
+ - 645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c
+ - 666: [improvement]: ARI debug should contain endpoint and method
+ - 676: [bug]: res_stir_shaken implicit declaration of function errors/warnings
+ - 689: [bug] Document the `Events` argument of the `Login` AMI action
+ - 713: [bug]: SNI isn't being set on websocket client connections
+ - 716: [bug]: Memory leak in res_stir_shaken tn_config, plus a few other issues
+ - 719: [bug]: segfault on start if compiled with DETECT_DEADLOCKS
+ - 783: [bug]: Under certain circumstances a channel snapshot can get orphaned in the cache
+ - 789: [bug]: Mediasec headers aren't sent on outgoing INVITEs
+ - ASTERISK-16799: Callee declined when 'beep' audio file does not exist
+ - ASTERISK-18069: [patch] app_queue Add Login Time and Last Paused Times to Queue Members
+ - ASTERISK-18416: [patch] Realtime queue agents unavailable via AMI before a call event.
+ - ASTERISK-18454: Option for Read to be able to accept #
+ - ASTERISK-20219: [patch] - IAX2 Call Encryption Fails with RSA authentication
+ - ASTERISK-20259: [patch] Update Doxygen Configuration for make progdocs
+ - ASTERISK-20339: chan_mgcp, resp_pktccops ast_debug support
+ - ASTERISK-21502: New SIP Channel Driver - add Advice of Charge support
+ - ASTERISK-21741: [patch] - Improved Caller ID Diagnostics and Processing for FXO Channels
+ - ASTERISK-21795: failed compilation - dns.c references res_nsearch which is not available on uclibc
+ - ASTERISK-22246: Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug)
+ - ASTERISK-24329: Music On Hold announcement cuts intro of music the first time it is played
+ - ASTERISK-24427: Documentation is missing for a few AMI Events - Including CDR and events triggered after the QueueStatus action
+ - ASTERISK-24434: Fix differing usage of assignment operators in modules.conf
+ - ASTERISK-24601: [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY event: dialog XML body
+ - ASTERISK-24631: Incorrect description of option "context" in queues.conf.sample
+ - ASTERISK-24827: Missing documentation for chan_dahdi dial string ring cadences
+ - ASTERISK-25716: Documentation: Document explanations and examples for possible values of DIALSTATUS
+ - ASTERISK-26424: app_voicemail: Undocumented behavior from VMSayName
+ - ASTERISK-26497: make install downloads x86_32 variants of external modules on non Intel architectures
+ - ASTERISK-26582: Asterisk seems to ignore the "n" parameter for "disable console colorization"
+ - ASTERISK-26614: app_queue: updatecdr option in queues.conf does effectively nothing
+ - ASTERISK-26689: res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity
+ - ASTERISK-26719: pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1)
+ - ASTERISK-26826: testsuite: Add support for Python 3
+ - ASTERISK-26894: pjsip should support tel uri scheme
+ - ASTERISK-26991: documentation: Doxygen site is no longer being updated
+ - ASTERISK-27176: test_abstract_jb: frames leak
+ - ASTERISK-27273: app_voicemail: When a voicemail is marked as "Urgent", it is not sent by email/processed by the mailcmd command
+ - ASTERISK-27406: Infinite loop when out of ports and rtpstart value is odd
+ - ASTERISK-27477: Chan_pjsip does not support unauthenticated OPTIONS ping
+ - ASTERISK-27542: app_queue: When "queue show" CLI command is executed a crash occurs
+ - ASTERISK-27597: AMI Queuestatus not working (with realtime queue)
+ - ASTERISK-27816: func_talkdetect's logic is completely broken
+ - ASTERISK-27830: Asterisk crashes on Invalid UTF-8 string
+ - ASTERISK-27871: Remote URL in playback must end with file extension
+ - ASTERISK-28004: dns: Core ast_dns_get_nameservers does not support configured IPv6 servers
+ - ASTERISK-28016: PJSIP sends duplicate 183 Progress responses
+ - ASTERISK-28040: pbx: "dialplan reload" is removing minus symbol from dynamic hints
+ - ASTERISK-28053: chan_pjsip: Wrong or missing Q.850 reason in CANCEL
+ - ASTERISK-28109: pbx_dundi: Does not support chan_pjsip
+ - ASTERISK-28185: chan_pjsip: Subsequent same responses are not stopped
+ - ASTERISK-28219: res_ari: Channel create and dial may cause "BUG! Must supply a channel name.." error
+ - ASTERISK-28233: pbx_dundi: PJSIP is not a supported technology
+ - ASTERISK-28237: "FRACK!, Failed assertion bad magic number" happens when unsubscribe an application from an event source
+ - ASTERISK-28311: dsp: ast_dsp_silence_noise_with_energy wrong judgment of frame format
+ - ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not working
+ - ASTERISK-28369: app_queue: Member device state "invalid" when second call is ringing and hint is used
+ - ASTERISK-28393: Multidomain support issue
+ - ASTERISK-28416: Unable to get rtp codec payload code for slin
+ - ASTERISK-28422: Memory Leak in Confbridge menu
+ - ASTERISK-28430: res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
+ - ASTERISK-28452: pjsip: <sess-version> of SDP is not incremented though SDP may be changed on reinvite without SDP offer
+ - ASTERISK-28518: chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold
+ - ASTERISK-28549: Two repeated 183
+ - ASTERISK-28689: res_pjsip: Crash when locking group lock when sending stateful response
+ - ASTERISK-28701: app_queue: Core reload resets queue stats, even when keepstats=yes
+ - ASTERISK-28767: chan_pjsip: Caller ID not used when checking for extension, callerid supplement executed too late
+ - ASTERISK-28798: [patch] chan_sip: TCP/TLS client without server.
+ - ASTERISK-28825: Any curl response checks out as valid even if 404 is returned.
+ - ASTERISK-28863: The ast_rtp_codecs_payloads functions don't preserve order
+ - ASTERISK-28878: chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16
+ - ASTERISK-28883: Spyee information ist missing in ChanSpyStop AMI Event
+ - ASTERISK-28890: res_pjsip_sdp_rtp: Keepalive not supported for video streams
+ - ASTERISK-28891: documentation: AGICommand_set+music documentation arguments displayed incorreclty
+ - ASTERISK-28927: Asterisk crash in music on hold
+ - ASTERISK-28933: res_pjsip.so fails to load when bundled pjproject is compiled without libssl
+ - ASTERISK-28947: Segmentation fault in mixmonitor_ds_destroy
+ - ASTERISK-28973: Malformed IP address in SDP of 2nd SIP timer triggered INVITE when NAT is active (UDP transport with external_media_address)
+ - ASTERISK-28974: res_rtp_asterisk: T.140 messages have appended RTP string to each message block.
+ - ASTERISK-28978: acl: named_acl rule misconfiguration results in segfault on reading rule from realtime
+ - ASTERISK-28987: BridgeCreated ARI event shows wrong video_mode info
+ - ASTERISK-28992: app_voicemail: Deadlock in ODBC when retrieving file
+ - ASTERISK-28995: res_pjsip_registrar: Expires on statically configured contacts is not correct
+ - ASTERISK-29001: chan_pjsip does not process or forward 181 responses
+ - ASTERISK-29011: chan_sip: ToHost property not cleared on reload
+ - ASTERISK-29013: res_pjsip: Asterisk doesn't stop sending invites (with auth) on 407 replies
+ - ASTERISK-29014: res_pjsip_session: Re-INVITE collisions aren't handled correctly
+ - ASTERISK-29021: [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified versions
+ - ASTERISK-29022: Crash when manipulating PJSIP invite dlg ref counts
+ - ASTERISK-29024: pjsip: Route Header in Cancel request incorrectly set
+ - ASTERISK-29027: Implement support for History-Info
+ - ASTERISK-29030: res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established
+ - ASTERISK-29033: res_pjsip_session: Aggressively terminates session on failed re-INVITE
+ - ASTERISK-29034: Lastpause of realtime members is reseting
+ - ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing
+ - ASTERISK-29040: res_speech: Assertion on format
+ - ASTERISK-29042: res_parking: Parker UUID is no longer copied
+ - ASTERISK-29043: app_queue: Leave empty sometimes not recorded as abandoned
+ - 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-29048: chan_iax2: "iax2 show registry" shows host for perceived
+ - ASTERISK-29051: res_pjsip_sdp_rtp: Does not set correct values on RTP instance when "auto" DTMF is used
+ - ASTERISK-29054: Logger: Add debug logging categories
+ - ASTERISK-29055: Create a Bridge with video_single mode
+ - ASTERISK-29056: Increase reg_server column size for ps_contacts table realtime
+ - ASTERISK-29057: pjsip: Crash on call rejection during high load
+ - ASTERISK-29071: app_confbridge: Memory rises when jitterbuffer enabled and muting over AMI occurs
+ - ASTERISK-29081: res_stasis: Add compare function for bridges moh container
+ - ASTERISK-29083: Do not build chan_sip by default as it is now deprecated
+ - ASTERISK-29085: func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT
+ - ASTERISK-29089: RTP Ports not cleared after hangup
+ - ASTERISK-29091: Crash when ast_translator_build_path fails
+ - ASTERISK-29097: res_pjsip_config_wizard: Crash when freeing string when failing to add extension
+ - ASTERISK-29099: res_musiconhold: Realtime MOH only loads a single entry
+ - ASTERISK-29105: chan_pjsip: 180 Ringing with SDP not changed into progress
+ - ASTERISK-29108: resource_endpoints.c : Memory leak if endpoint not found
+ - ASTERISK-29109: res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
+ - ASTERISK-29118: VoiceMail() should have an option to play greetings as Early Media
+ - ASTERISK-29123: logger.conf.sample missing comment mark on line 115
+ - ASTERISK-29124: res_pjsip: flow transport broken for outbound requests
+ - ASTERISK-29130: prometheus: Crash when scraping bridge
+ - ASTERISK-29136: config: Sample features.conf incorrectly includes " around sound files
+ - ASTERISK-29142: sip_to_pjsip.py: doesn't read globbed includes
+ - ASTERISK-29143: res_http_media_cache: HTTP media cache stored hardcoded in /tmp
+ - ASTERISK-29144: GCC Warnings with OPTIMIZE=-Og make
+ - ASTERISK-29145: GCC Warnings with OPTIMIZE=-Os make
+ - ASTERISK-29146: GCC Warnings: ‘%s’ directive argument is null.
+ - ASTERISK-29148: AST_MODULE_INFO no, MODULEINFO depend
+ - ASTERISK-29155: app_queue: Deadlock between queues container and individual queues
+ - ASTERISK-29161: Incorrect setup of recall channels
+ - ASTERISK-29165: res_pjsip: malformed header Accept-Encoding in OPTIONS response
+ - ASTERISK-29168: Asterisk crashes during call transfer
+ - ASTERISK-29173: Media cache URL requests allow infinite redirects
+ - ASTERISK-29175: res_pjsip_stir_shaken: Fix module description
+ - ASTERISK-29185: chan_pjsip: Endpoint: allow = all is broken.
+ - ASTERISK-29188: null media causing the Asterisk crash
+ - ASTERISK-29191: tel: URI in Diversion header causes crash
+ - ASTERISK-29196: res_pjsip: Segmentation fault
+ - ASTERISK-29201: Crash occurs when Transfer and execute Hangup before the Transfer result
+ - ASTERISK-29203: res_pjsip_t38: Crash when changing state
+ - ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client
+ - ASTERISK-29209: Debug messages printed by scope trace might be missing newlines
+ - ASTERISK-29210: res_pjsip: Crash when examining transport
+ - ASTERISK-29211: res_musiconhold: Segfault on realtime music on hold without entries
+ - ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused asterisk crash
+ - ASTERISK-29216: contrib: systemd asterisk service for centos8 or other newer linux versions
+ - ASTERISK-29217: LOCK() can grant the same lock to multiple channels spuriously
+ - ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains History-Info
+ - ASTERISK-29220: After T38 reinvite response of 488 a subsequent G711 reinvite is not processed correctly. Instead the previous T38 session media is used
+ - ASTERISK-29222: chan_sip: Hold/Resume an sRTP call on a video enabled user-agent.
+ - ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash
+ - ASTERISK-29229: Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription
+ - ASTERISK-29230: pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send
+ - ASTERISK-29231: pjsip: SIGSEGV in CLI if no trunk is registered
+ - ASTERISK-29235: res_pjsip_nat: Contact is rewritten on REGISTER responses with external_signaling_address
+ - ASTERISK-29237: chan_sip: SDP: m=video is parsed even when disabled.
+ - ASTERISK-29238: chan_sip: SDP: Offers without any enabled stream are accepted.
+ - ASTERISK-29240: chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable
+ - ASTERISK-29241: pjsip / register: wrong port used in Contact and Via if multiple transports are defined.
+ - ASTERISK-29244: Add MixMonitorStart / Stop / Mute AMI events
+ - ASTERISK-29248: res_pjsip_session: res sometimes uninitialized reported by compiler Clang.
+ - ASTERISK-29252: TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER) failure SIP code
+ - ASTERISK-29253: Incorrect bridging on transfer
+ - ASTERISK-29258: chan_sip: Audio stream rejected, Other stream present: Invalid SDP.
+ - ASTERISK-29259: channel: Allow text+video media streams, again.
+ - ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
+ - ASTERISK-29261: res_pjsip: user=phone validation fail for isup numbers containing *#
+ - ASTERISK-29262: Support of various URL-schemes by MoH
+ - ASTERISK-29265: chan_sip: Allow text+video media streams, again.
+ - ASTERISK-29266: ICE Role conflict with an unauthorized session
+ - ASTERISK-29275: Support of MIME-type for wav16
+ - ASTERISK-29280: chan_sip: Allow peers without audio (text+video).
+ - ASTERISK-29287: app.h: C++ compatibility broken
+ - ASTERISK-29293: res_config_pgsql: Limit realtime_pgsql() to return one (no more) record
+ - ASTERISK-29297: say: Y2021 problem – Asterisk cannot say year 2021 in Dutch
+ - ASTERISK-29300: res_rtp_asterisk: When native local bridging the remote SSRC becomes permanent
+ - ASTERISK-29303: pjsip: Re-invite occurs when it shouldn't
+ - ASTERISK-29305: ASTERISK-29203 / AST-2021-002 -- Another scenario is causing a crash
+ - ASTERISK-29306: strings: Incorrect use of __attribute__((pure)) in ast_str_to_lower definition
+ - ASTERISK-29311: res_odbc_transaction sets forcecommit default value based on isolation level instead of forcecommit
+ - ASTERISK-29312: res_fax: asterisk fails to publish the Stasis and ReceiveFax status messages if the remote Station ID contains invalid UTF-8 characters
+ - ASTERISK-29313: res_pjsip_refer: Segfault in progress notify
+ - ASTERISK-29315: res_pjsip: re-registration gets stuck if setting initial auth credentials fails
+ - ASTERISK-29320: res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold
+ - ASTERISK-29321: sorcery: Add support for more intelligent reloading.
+ - ASTERISK-29325: res_pjsip_registrar: Include source IP address and port in log messages
+ - ASTERISK-29326: asterisk: Update copyright/company
+ - ASTERISK-29328: translate.c: possible buffer overflow when upsampling
+ - ASTERISK-29329: app_dial: DTMF to 'D' option gets duplicated if there are multiple progress events
+ - ASTERISK-29335: xml: Embed module information into core XML documentation.
+ - ASTERISK-29336: documentation: Fix inconsistent support levels
+ - ASTERISK-29337: menuselect: Add ability to set deprecated in and removed in versions for modules
+ - ASTERISK-29339: loader: Let's output warnings for deprecated modules!
+ - ASTERISK-29348: menuselect doesn't return errors in many cases
+ - ASTERISK-29349: Silent voicemail option is not completely silent
+ - ASTERISK-29351: Qualify pjproject 2.12 for Asterisk
+ - ASTERISK-29352: res_rtp_asterisk: Fix frame delivery time when SSRC changes
+ - ASTERISK-29353: Qualify jansson 2.14 for asterisk
+ - ASTERISK-29354: res_pjsip: Allow partial reloading of transports
+ - ASTERISK-29355: app_queue: Queue member status message sent even if status doesn't change
+ - ASTERISK-29358: chan_pjsip: Trace message for progress is output even if frame is not queued
+ - ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation
+ - ASTERISK-29365: taskprocessor: Can cause assert at shutdown
+ - ASTERISK-29370: chan_sip does not recognize application/hook-flash
+ - ASTERISK-29372: file.c switch does not account for flash events
+ - ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated
+ - ASTERISK-29377: cpool_release_pool "double free or corruption (out)"
+ - ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at channel_internal_api.c:1590
+ - ASTERISK-29380: Add Flash AMI event to handle flash events
+ - ASTERISK-29381: chan_pjsip: Remote denial of service by an authenticated user
+ - ASTERISK-29389: Add PJSIP_HEADERS() and ability to read header by pattern
+ - ASTERISK-29391: VoiceMail does not cancel recording on rerecord hangup
+ - ASTERISK-29392: chan_iax2: Asterisk crashes when queueing video with format
+ - ASTERISK-29397: pjsip: Asterisk isn't tolerant of RFC8760 UASs
+ - ASTERISK-29402: res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform does not support it
+ - ASTERISK-29404: Consolidate res_pjsip_messaging fixes for domain name
+ - ASTERISK-29407: chan_local: Filtering audio formats should not occur on removed streams
+ - ASTERISK-29411: Crash in pjsip_msg_find_hdr_by_name
+ - ASTERISK-29415: Crash in PJSIP TLS transport
+ - ASTERISK-29428: DTMF on progress results in infinite loop if progress followed by hangup received
+ - ASTERISK-29431: Minimum and maximum dialplan functions
+ - ASTERISK-29432: New function to allow access to any channel
+ - ASTERISK-29433: res_rtp_asterisk: Server reflexive candidates use incorrect raddr for RTCP
+ - ASTERISK-29434: Asterisk reveals pjproject version in STUN packets
+ - ASTERISK-29439: func_volume: Volume function can't be read
+ - ASTERISK-29440: app_confbridge: Allow ConfBridge answer to be suppressed
+ - ASTERISK-29441: Core reload making TCP endpoints go offline
+ - ASTERISK-29442: app_dial: Expand A option to allow announcement playback to caller
+ - ASTERISK-29444: Add application to wait for condition
+ - ASTERISK-29446: app_confbridge: New ConfKick application
+ - ASTERISK-29450: Allow setting channel variables using Originate application
+ - ASTERISK-29453: alembic: incoming_call_offer_pref and outgoing_call_offer_pref missing in "ps_endpoints" table
+ - ASTERISK-29454: New application to reload modules
+ - ASTERISK-29459: Missing configuration from PJSIP to SIP conversion script
+ - ASTERISK-29460: Recognize application/hook-flash in PJSIP
+ - ASTERISK-29464: ARI - PlaybackFinish skip error events
+ - ASTERISK-29472: res_pjsip: OLI/ANI2 support missing
+ - ASTERISK-29475: SayNumber triggers WARNING if caller hangs up during application execution
+ - ASTERISK-29476: res_stir_shaken: Blind SSRF vulnerabilities
+ - ASTERISK-29477: Function to asynchronously store digits dialed
+ - ASTERISK-29478: Function to drop frames in the TX or RX directions
+ - ASTERISK-29479: [patch] Channels are not put on hold for Session Progress with inactive audio
+ - ASTERISK-29480: fixedjitterbuffer contains an un-wrappered assert that triggers on a negative time slew
+ - ASTERISK-29485: core: Inband generation of tones for Busy() and Congestion() may not occur
+ - ASTERISK-29486: Hint-like extension value lookup function without device state
+ - ASTERISK-29494: cdr_adaptive_odbc: Prevent throwing warnings if CDR filtering is used
+ - ASTERISK-29495: Return integer instead of float if response is a whole number
+ - ASTERISK-29496: Add SendMF application
+ - ASTERISK-29497: Add conditional branch applications
+ - ASTERISK-29501: ARI - Stasis Playback doesn't hangup call when processing a list of invalid files
+ - ASTERISK-29503: Updated identify/match syntax not supported by config wizard
+ - ASTERISK-29507: STUN timeout is silently delaying calls
+ - ASTERISK-29508: STUN server address refresh
+ - ASTERISK-29513: statsd: Remove non-standard metric type Meter
+ - ASTERISK-29514: ari: Audiosocket segfault when no data specified
+ - ASTERISK-29515: app_queue: QueueSummary and QueueStatus events don't exist in documentation
+ - ASTERISK-29516: app_senddtmf / local: Sending DTMF does not work when not answered
+ - ASTERISK-29518: sig_analog: FCG_CAMA fails to signal ANI spill when using MF signaling
+ - ASTERISK-29525: PJSIP remove_existing unavailable contacts
+ - ASTERISK-29526: G729 audio gets corrupted by Asterisk due to smoother
+ - ASTERISK-29527: res_http_media_cache: Cleanup audio format lookup in HTTP requests
+ - ASTERISK-29528: Add support for multiple files for agent announcements
+ - ASTERISK-29529: Add custom logging level
+ - ASTERISK-29531: Add SAYFILES function
+ - ASTERISK-29535: Segmentation fault in libasteriskpj.so.2
+ - ASTERISK-29539: Segmentation fault at ast_writestream() when write handler not defined (happens with OGG/Speex)
+ - ASTERISK-29540: aelparse: include of context with timings fails
+ - ASTERISK-29541: app_morsecode: Add American Morse code
+ - ASTERISK-29542: Add audio scrambler
+ - ASTERISK-29543: app_originate: Allow specifying codec(s) to use
+ - ASTERISK-29544: Media Cache - Delayed remote sound file retrieve delays all playbacks
+ - ASTERISK-29546: Add tone detection module
+ - ASTERISK-29548: app_meetme: Deprecated in 19, to be removed in 21
+ - ASTERISK-29549: app_osploop: Deprecated in 19, to be removed in 21
+ - ASTERISK-29550: chan_alsa: Deprecated in 19, to be removed in 21
+ - ASTERISK-29551: chan_mgcp: Deprecated in 19, to be removed in 21
+ - ASTERISK-29552: chan_skinny: Deprecated in 19, to be removed in 21
+ - ASTERISK-29553: res_pktccops: Deprecated in 19, to be removed in 21
+ - ASTERISK-29558: app_macro: Deprecated in 16, to be removed in 21
+ - ASTERISK-29567: chan_sip: Deprecated in 17, to be removed in 21
+ - ASTERISK-29572: res_monitor: Deprecated in 16, to be removed in 21
+ - ASTERISK-29575: app_milliwatt: Milliwatt application doesn't use the proper timings
+ - ASTERISK-29578: app_queue: Custom device state using included hints do not update
+ - ASTERISK-29582: res_pjproject: Can't map pjproject log messages to Asterisk TRACE
+ - ASTERISK-29584: cdr_mysql: Remove deprecated module
+ - ASTERISK-29585: app_mysql: Remove deprecated module
+ - ASTERISK-29586: app_ices: Remove deprecated module
+ - ASTERISK-29587: app_fax: Remove deprecated module
+ - ASTERISK-29588: app_url: Remove deprecated module
+ - ASTERISK-29589: app_image: Remove deprecated module
+ - ASTERISK-29590: app_nbscat: Remove deprecated module
+ - ASTERISK-29591: app_dahdiras: Remove deprecated module
+ - ASTERISK-29592: cdr_syslog: Remove deprecated module
+ - ASTERISK-29593: chan_oss: Remove deprecated module
+ - ASTERISK-29594: chan_phone: Remove deprecated module
+ - ASTERISK-29595: chan_nbs: Remove deprecated module
+ - ASTERISK-29596: chan_misdn: Remove deprecated module
+ - ASTERISK-29597: chan_vpb: Remove deprecated module
+ - ASTERISK-29598: res_config_sqlite: Remove deprecated module
+ - ASTERISK-29599: conf2ael: Remove deprecated application
+ - ASTERISK-29600: muted: Remove deprecated application
+ - ASTERISK-29601: moduleinfo: Add replacement module information
+ - ASTERISK-29602: res_monitor: Disable building by default.
+ - ASTERISK-29603: res_pjsip: UPDATE/re-INVITE not sent when "timers=always" is specified in pjsip.conf
+ - ASTERISK-29604: ari: Segfault with lots of calls
+ - ASTERISK-29605: chan_iax2: Add ANI2
+ - ASTERISK-29609: Subsequent 'ael reload' will cause a lock up
+ - ASTERISK-29612: bridge_basic: Don't throw warning if attended transfer is cancelled
+ - ASTERISK-29614: app_agent_pool: XML Doc: unterminated entity reference
+ - ASTERISK-29616: res_rtp_asterisk: sqrt(.) requires the header math.h.
+ - ASTERISK-29618: ConfBridge errors on creation conference room
+ - ASTERISK-29622: ARI: external media create doesn't use body parameter
+ - ASTERISK-29624: Contact identifier is not updated when FDQN resolves to a new address
+ - ASTERISK-29625: srtp cryptos accepted if not enabled
+ - ASTERISK-29626: app_stack: Include calling location if attempting to branch to nonexistent location
+ - ASTERISK-29627: Add STRBETWEEN function
+ - ASTERISK-29628: Add file and directory functions
+ - ASTERISK-29629: ARI external media channel creation doesn't set option data
+ - ASTERISK-29630: Asterisk is unable to read extended number format terminfo files
+ - ASTERISK-29632: Add option to Application_VoiceMail to suppress instructions only when a custom greeting is present
+ - ASTERISK-29634: res_snmp: gcc 11 needs -fPIC to compile correctly
+ - ASTERISK-29635: MP3Player don' t work with actual mpg123 versions
+ - ASTERISK-29637: Add support for future dates in Say.c
+ - ASTERISK-29638: res_pjsip_session: No video after early media
+ - ASTERISK-29654: pjproject includes trailing whitespace in sdp format attributes
+ - ASTERISK-29655: res_pjsip_session: No video to caller if no camera available
+ - ASTERISK-29656: Add CHANNEL_EXISTS function
+ - ASTERISK-29660: Build failure when disabling PJSIP support
+ - ASTERISK-29661: func_vmcount: Add support for multiple mailboxes
+ - ASTERISK-29662: Add mix option to Playback application for say and filename
+ - ASTERISK-29663: messaging: AMI MessageSend does not support same parameters as dialplan application
+ - ASTERISK-29664: PJSIP processing token with % incorrectly
+ - ASTERISK-29668: ari: Listing bridges fails when dialing bridge exists
+ - ASTERISK-29671: res_rtp_asterisk: memory leak
+ - ASTERISK-29673: app_read: Fix null pointer crash regression
+ - ASTERISK-29674: Adjust for 64bit time_t
+ - ASTERISK-29682: Squash compiler issues generated by gcc 11
+ - ASTERISK-29691: stun: Not all users provide a dst to ast_stun_request
+ - ASTERISK-29693: Using --with-crypto and --with-ssl fails on a recompile
+ - ASTERISK-29695: SAY.CONF wrong logic when converting 24hour time to say 12 hour am/pm
+ - ASTERISK-29698: Segfault if sorcery object_lifetime_maximum and qualify_frequency the same value
+ - ASTERISK-29702: sig_analog: Fix truncated buffer copy
+ - ASTERISK-29703: res_pjsip_callerid: Fix OLI parsing
+ - ASTERISK-29705: app_read: Fix custom terminator functionality regression
+ - ASTERISK-29706: func_json: Add JSON parsing function
+ - ASTERISK-29707: chan_iax2: Allow both key and secret to be specified at dial time
+ - 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-29714: Spelling errors
+ - ASTERISK-29715: app_voicemail: Refactor email generation functions
+ - ASTERISK-29717: res_config_sqlite: not removed in makeopts.in
+ - ASTERISK-29720: res_tonedetect: Add call progress tone detection
+ - ASTERISK-29722: test_timezone_watch breaks during DST to ST transition
+ - ASTERISK-29724: BuildSystem: In POSIX sh, == in place of = is undefined.
+ - ASTERISK-29726: Add Asterisk External Application Protocol (AEAP) implementation
+ - ASTERISK-29727: Add type for JSON stasis message RTCP Report Received/Sent
+ - ASTERISK-29728: menuselect: Disabled by default modules that are enabled are always recompiled
+ - ASTERISK-29729: Incompatibility with newer spandsp releases (3.0.0+)
+ - ASTERISK-29730: Segfault in __ao2_ref if refdebug = yes
+ - ASTERISK-29732: progdocs: Fix grouping for latest Doxygen
+ - ASTERISK-29733: progdocs: Avoid name with Doxygen \file
+ - ASTERISK-29734: progdocs: Use Doxygen \example correctly
+ - ASTERISK-29735: progdocs: Avoid multiple use of section labels
+ - ASTERISK-29736: bridge_channel: Fix for Doxygen
+ - ASTERISK-29737: chan_iax2: Fix for Doxygen
+ - ASTERISK-29740: apps: Fix for Doxygen
+ - ASTERISK-29741: tests: Fix for Doxygen
+ - ASTERISK-29742: addons: Fix for Doxygen.
+ - ASTERISK-29743: bridges: Fix for Doxygen
+ - ASTERISK-29744: app_morsecode: Fix deadlock
+ - ASTERISK-29745: pbx: Add public API for more elegant variable substitution with extensions
+ - ASTERISK-29746: tcptls.c: TCP client connect fails due to interrupt
+ - ASTERISK-29747: res_pjsip: Fix for Doxygen
+ - ASTERISK-29748: bridging: Infinite loop when both Local channel halves in same bridge
+ - ASTERISK-29749: res_xmpp: Fix for Doxygen
+ - ASTERISK-29750: stasis: Fix for Doxygen
+ - ASTERISK-29751: channel: Fix for Doxygen
+ - ASTERISK-29752: app: Fix for Doxygen
+ - ASTERISK-29753: parking: Fix for Doxygen
+ - ASTERISK-29754: odbc: Fix for Doxygen
+ - ASTERISK-29755: frame: Fix for Doxygen
+ - ASTERISK-29756: res_ari: Fix for Doxygen
+ - ASTERISK-29758: configs: Minor updates to sample configs
+ - ASTERISK-29759: app_sendtext: Add ReceiveText application
+ - ASTERISK-29761: res: Fix for Doxygen
+ - ASTERISK-29762: channels: Fix for Doxygen
+ - ASTERISK-29763: main: Fix for Doxygen
+ - ASTERISK-29765: xmldoc: Fix for Doxygen
+ - ASTERISK-29766: pbx_variables: MSet truncates sets after 24 variables
+ - ASTERISK-29771: Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning
+ - ASTERISK-29772: chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause a coredump
+ - ASTERISK-29773: progdocs: doxyref.h outdated
+ - ASTERISK-29776: stir/shaken: Requires GNU designator
+ - ASTERISK-29777: documentation: Standardize example syntax
+ - ASTERISK-29779: progdocs: Hidden code sections with syntax errors.
+ - ASTERISK-29785: res_pjsip_sdp_rtp: Warns on every offered crypto suite
+ - ASTERISK-29790: xmldoc: Dump invalid to XML DTD: XSLT
+ - ASTERISK-29791: xmldoc: Dump invalid to XML DTD: ACO Matchfield
+ - ASTERISK-29793: adsi: CAS is malformed
+ - ASTERISK-29794: ast_coredumper does not delete results when requested and a specific output dir is set
+ - ASTERISK-29795: DIALEDPEERNUMBER not set on destination channel for Queue calls
+ - ASTERISK-29797: Support for Danish language syntax in VM
+ - ASTERISK-29800: strings: Fix misusage in comment examples
+ - ASTERISK-29801: app.c: Throw warnings for nonexistent options
+ - ASTERISK-29802: app_sf: Add full tech-agnostic SF support
+ - ASTERISK-29803: pbx_variables: cp4 variables is used uninitialized
+ - ASTERISK-29804: bundled_pjproject: sip_inv is missing multipart support in some cases
+ - ASTERISK-29806: app_queue: extension state incorrect
+ - ASTERISK-29807: cli: add module refresh command
+ - ASTERISK-29808: cdr: allow disabling CDR by default
+ - ASTERISK-29809: curl, stir_shaken: refactor curl code
+ - ASTERISK-29810: app_signal: Add channel signaling applications
+ - ASTERISK-29813: res_pjsip_session doesn't support multipart message bodies
+ - ASTERISK-29815: dsp: Define magic number as macro
+ - ASTERISK-29816: SAY_DTMF_INTERRUPT channel variable is not honored
+ - ASTERISK-29817: gethostbyname_r is misdetected on NetBSD and causes a build failure
+ - ASTERISK-29818: Build failure on NetBSD due to hmac function collision
+ - ASTERISK-29819: utils.c: Remove all usages of ast_gethostbyname()
+ - ASTERISK-29820: cli: Add command to evaluate a function
+ - ASTERISK-29821: Deadlock in bridge_channel_internal_join() on local channels.
+ - ASTERISK-29822: cli: Typing \? freezes the CLI permanently with remote console
+ - ASTERISK-29824: It's hard to make changes to bundled pjproject
+ - ASTERISK-29827: Support for Nordic language syntax in Queues
+ - ASTERISK-29829: app_mp3: Throw warning if attempting to play a nonexistent stream
+ - ASTERISK-29830: ami: Add AMI event for Wink
+ - ASTERISK-29831: Queue don't play "thank-you" when here is no hold time announcements
+ - ASTERISK-29832: Enable pickup on channel after having received 183 Progress
+ - ASTERISK-29838: ${SQL_ESC()} not correctly escaping a terminating \
+ - ASTERISK-29840: func_channel: Add LASTCONTEXT and LASTEXTEN fields
+ - ASTERISK-29842: Do not change 180 Ringing to 183 Progress even if early_media already enabled
+ - ASTERISK-29843: Session timers get removed on UPDATE
+ - ASTERISK-29845: res_pjsip_outbound_registration: Show time remaining until registration lapses
+ - ASTERISK-29846: channels: bad ao2 ref causes crash
+ - ASTERISK-29847: pbx_variables: ASTSBINDIR is missing
+ - ASTERISK-29848: documentation: Document special system and channel variables
+ - ASTERISK-29850: ast_get_tid() not implemented for NetBSD
+ - ASTERISK-29851: rdtsc is not enabled (stubbed out) on NetBSD
+ - ASTERISK-29852: make_version uses GNU-ism that break git-svn-id parsing on NetBSD
+ - ASTERISK-29853: ami: Allow events to be globally disabled
+ - ASTERISK-29854: func_frame_drop: fix buffer usage typo
+ - ASTERISK-29855: frame.h: fix CNG documentation typo
+ - ASTERISK-29856: res_rtp_asterisk: Invalid comparison creates unreachable code
+ - ASTERISK-29857: res_tonedetect: fix logic errors in code
+ - ASTERISK-29858: Regression: Using external pjproject not working after "hack" commit
+ - ASTERISK-29859: VoiceMailMain() fails when encountering non-numeric CALLERID(num)
+ - ASTERISK-29861: asterisk.h: add macro for curl user agent
+ - ASTERISK-29866: cli: add core dump information to core show settings
+ - ASTERISK-29867: configure fails if libsrtp dev files are not installed
+ - ASTERISK-29869: rtp sequence number can skip after DTMF under certain bridges
+ - ASTERISK-29871: res_prometheus: Failure to load causes FRACKs
+ - ASTERISK-29872: res_stir_shaken: Resource exhaustion with large files
+ - ASTERISK-29873: [patch] Queue Realtime load
+ - ASTERISK-29876: app_queue: Add music on hold option
+ - ASTERISK-29877: app_mf: Allow reading a maximum number of digits
+ - ASTERISK-29886: Asterisk AMI sends not-valid XML
+ - ASTERISK-29888: res_pjsip_outbound_authenticator_digest: ABRT attempting to clean up auth_sess
+ - ASTERISK-29891: [patch] provide a display name for RLS subscriptions
+ - ASTERISK-29895: chan_iax2: Fix misaligned spacing in iax2 show netstats printout
+ - ASTERISK-29896: xmldocs: Add since tag
+ - ASTERISK-29897: channels: Increase core debug levels for chatty debugs
+ - ASTERISK-29898: documentation: Add default attributes to documentation
+ - ASTERISK-29899: features: Add advanced transfer initiation options
+ - ASTERISK-29900: app_mp3: Document and warn about https incompatibility
+ - ASTERISK-29904: RLS: Batched Notifications stop working
+ - ASTERISK-29905: OSX: bininstall launchd issue on cross-platfrom build
+ - ASTERISK-29906: [patch] update RLS to reflect the changes to the lists
+ - ASTERISK-29907: res_pjsip, app_confbridge: Video call through ConfBridge with normal endpoints causes infinite loop/crash
+ - ASTERISK-29909: app_queue: Add support for withdrawing a call
+ - ASTERISK-29913: func_json: Adds multi-level and array parsing to JSON_DECODE
+ - ASTERISK-29917: ami: FilterList action doesn't exist
+ - ASTERISK-29920: app_voicemail: Warn if trying to manage nonexistent mailbox
+ - ASTERISK-29923: docs, LICENSE: pbx.digium.com no longer exists
+ - ASTERISK-29924: res_config_pgsql: omit "unsupported column type 'text'" error
+ - ASTERISK-29925: func_db: Warn about malformed key names
+ - ASTERISK-29928: logging messages truncated when using MUSL runtime
+ - ASTERISK-29929: res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions
+ - ASTERISK-29931: Option to allow a user to not hear the join sound on enter but everyone else can
+ - ASTERISK-29935: build: Remove leftover build references
+ - ASTERISK-29939: agi: Fix xmldoc bug with set music
+ - ASTERISK-29940: general: Add since tags to xmldocs
+ - ASTERISK-29941: chan_pjsip: Add ability to send flash events
+ - ASTERISK-29943: file.c: seeking to negative file offset is not prevented
+ - ASTERISK-29948: iostream: Infinite TCP timeout writing data
+ - ASTERISK-29950: SayNumber can handle '01' to '07', but not '08' or '09'
+ - ASTERISK-29951: app_mf, app_sf: Return -1 on hangup
+ - ASTERISK-29954: app_meetme: Emit warning if conference not found
+ - ASTERISK-29955: chan_sip: SIP route header is missing on UPDATE
+ - ASTERISK-29960: ari: Retrieving stored recording can returns wrong file
+ - ASTERISK-29961: RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request
+ - ASTERISK-29965: res_pjsip_outbound_registration: Make max registration delay configurable
+ - ASTERISK-29966: pbx_variables: ast_str_strlen can be wrong
+ - ASTERISK-29967: pbx_builtins: Add missing documentation
+ - ASTERISK-29968: func_db: Add a function to return cardinality of keys at prefix
+ - ASTERISK-29970: Use pkg-config to find libxml2 headers and libraries
+ - ASTERISK-29976: Should Readme include information about install_prereq script?
+ - ASTERISK-29980: build: External binary modules don't use https
+ - ASTERISK-29981: res_calendar: Asterisk crashes when starting, and will not run
+ - 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-29989: app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
+ - ASTERISK-29990: chan_dahdi: adding ring cadences is not idempotent on dahdi restart
+ - ASTERISK-29991: chan_dahdi, callerid: Caller ID does not honor presentation
+ - ASTERISK-29992: chan_dahdi: Allow pulse and tone dialing to be disabled
+ - ASTERISK-29993: chan_dahdi: Operator control option borks both lines involved on callee disconnect
+ - ASTERISK-29994: chan_dahdi: Round robin array size is too small for max number of groups
+ - ASTERISK-29998: sla: deadlock when calling SLAStation application
+ - ASTERISK-29999: pjsip: Get information from 200 OK INVITE reply headers
+ - ASTERISK-30000: chan_dahdi: Add POLARITY function
+ - ASTERISK-30001: db: Removing nonexistent entries shows "Database entry removed"
+ - ASTERISK-30002: app_meetme: Don't erroneously set global variables when channel is NULL
+ - ASTERISK-30003: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes
+ - ASTERISK-30004: chan_dahdi: Allow flash to hold to time out to silence
+ - ASTERISK-30006: res_pjsip: UDP transport does not work when async_operations is greater than 1
+ - ASTERISK-30007: chan_iax2: Prevent crashes due to attempted encryption with missing secrets
+ - ASTERISK-30008: samples: Remove obsolete config files
+ - ASTERISK-30013: core_local: Local channels cannot have slashes in the destination
+ - ASTERISK-30015: pjsip / WebRTC: Chrome creating large number of SDP attributes
+ - ASTERISK-30018: app_meetme: MeetmeList AMI event not documented
+ - ASTERISK-30020: ConfbridgeListRooms Event Not Documented
+ - ASTERISK-30021: ast_variable_list_replace_variable uses variable with new keyword
+ - ASTERISK-30023: cdr_adaptive_odbc: does not support DATETIME database columns
+ - ASTERISK-30024: Failed to sign STIR/SHAKEN payload with functionality not enabled
+ - ASTERISK-30027: ari: expose channel driver's unique id (i.e. Call-ID for chan_sip/chan_pjsip) in ARI channel resource
+ - ASTERISK-30029: build: Git security vulnerability fix is sad with our accessing git as root during "make install"
+ - ASTERISK-30032: Support of mediasec SIP headers and SDP attributes
+ - ASTERISK-30036: app_confbridge: Add CONFBRIDGE_CHANNELS function
+ - ASTERISK-30037: Add test support to calling external processes
+ - ASTERISK-30039: cli: Targeted debug on startup deadlocks and creates unstable system
+ - ASTERISK-30042: res_pjsip_transport_websocket: Registration over websocket returns a rewritten contact
+ - ASTERISK-30043: Wrong party is disconnected when hook-flashing on 3-way bridge
+ - ASTERISK-30044: GCC 12 issues
+ - ASTERISK-30045: Add test coverage to res/res_crypto.c functionality
+ - ASTERISK-30046: Reimplement res/res_crypto.c internals with EVP_PKEY interface to Openssl API's
+ - ASTERISK-30050: Upgrade Asterisk to bundled pjproject 2.12.1
+ - ASTERISK-30051: res_pjsip: No video after un-hold with moh_passthrough=yes
+ - ASTERISK-30058: Evaluate dialplan functions and variables in agi exec
+ - ASTERISK-30059: menuselect: libxml include fails under Gentoo
+ - ASTERISK-30060: loader: format warnings in dev mode
+ - ASTERISK-30061: pbx: Add pbx helper application
+ - ASTERISK-30062: cli: Add CLI command to execute a dialplan app
+ - ASTERISK-30063: app_voicemail: Add option to prevent deletion of messages
+ - ASTERISK-30064: pbx: iax2 switch causes crash due to deadlock and assertion
+ - ASTERISK-30065: pjsip: Open Websocket connection is not reused for outgoing requests
+ - ASTERISK-30072: res_pjsip: allow TLS verification of wildcard cert-bearing servers
+ - ASTERISK-30075: say: Abort if channel hangs up during playback
+ - ASTERISK-30076: app_stack: Incorrect exit location in predial handlers logged
+ - ASTERISK-30083: chan_iax2: Optional dependency on openssl/res_crypto is now mandatory
+ - ASTERISK-30086: res_parking: Warn when invalid parking space requested
+ - ASTERISK-30087: res_parking: Add music on hold override option
+ - ASTERISK-30089: general: fix typos
+ - ASTERISK-30090: xmldocs: Use example tags for examples
+ - ASTERISK-30091: cdr: Allow CDRs to ignore call state changes
+ - ASTERISK-30092: DateTime application: wrong inflection for one o'clock in German
+ - ASTERISK-30096: cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is unsupported at this time
+ - ASTERISK-30097: console: Recent documentation changes for connecting to remote console are inconsistent
+ - ASTERISK-30099: test_aeap_transport: transport_connect_fail sporadically causes failure
+ - ASTERISK-30100: res_pjsip: Path is ignored on INVITE to endpoint
+ - ASTERISK-30101: res_prometheus: Optional load res_pjsip_outbound_registration.so
+ - ASTERISK-30103: chan_ooh323 Vulnerability in calling/called party IE
+ - ASTERISK-30106: res_calendar_icalendar: Microsoft online ICS calendars no longer work
+ - ASTERISK-30107: iostream: Build failure with libressl
+ - ASTERISK-30109: res_pjsip: no contact-status AMI event on register of prune-on-boot contact that uses the same URI as before Asterisk restart
+ - ASTERISK-30115: app_dial: Allow hook flashes to propogate on outbound dials
+ - ASTERISK-30117: pbx_lua: Remove compiler warnings
+ - ASTERISK-30123: features: Update automixmon documentation to reflect reality
+ - ASTERISK-30126: Spelling mistake in configs/samples/queues.conf.sample
+ - ASTERISK-30127: Create core Geolocation capability for Asterisk
+ - ASTERISK-30128: Create PJSIP interface module for Geolocation
+ - ASTERISK-30135: [res_musiconhold] Allows the moh only for the answered call
+ - ASTERISK-30136: db: Add AMI action to retrieve all keys beginning with a prefix
+ - ASTERISK-30137: manager: Global disabled event filtered is incomplete
+ - ASTERISK-30138: Compile failure in res_geolocation/geoloc_eprofile.c when optimization is enabled
+ - ASTERISK-30143: manager: Read and Write output from "manager show connected" is not well documented/useful
+ - ASTERISK-30146: res_pjsip_logger: Add method-based log filtering
+ - ASTERISK-30150: res_pjsip_session: Add support for custom parameters
+ - ASTERISK-30151: Documentation doesn't include info about "field", a 3rd required parameter.
+ - ASTERISK-30153: logger: Improve log levels
+ - ASTERISK-30158: PJSIP: Add new 100rel option "peer_supported"
+ - ASTERISK-30159: general: Remove obsolete SVN references
+ - ASTERISK-30160: cdr.conf: Remove obsolete app_mysql reference
+ - ASTERISK-30161: locks: add AMI event for deadlock
+ - ASTERISK-30162: when chan_iax is used to relay calls, no ringing indication is played
+ - ASTERISK-30163: general: fix minor formatting issues
+ - ASTERISK-30164: chan_iax2: Add missing option documentation
+ - ASTERISK-30167: res_geolocation: Refactor for issues found by users
+ - ASTERISK-30176: manager: GetConfig can read files outside of Asterisk
+ - ASTERISK-30177: res_geolocation: Add option to suppress empty elements
+ - ASTERISK-30178: extend user_eq_phone behavior to local uri's
+ - ASTERISK-30179: app_amd: Allow audio to be played while AMD is running
+ - ASTERISK-30180: app_broadcast: Add a channel audio multicasting application
+ - ASTERISK-30182: res_geolocation: Add built-in profiles to use in fully dynamic configurations
+ - ASTERISK-30184: res_pjsip_session: re-INVITE after answering results in wrong stream direction of first call leg
+ - ASTERISK-30185: res_geolocation: Allow location parameters to be specified in profiles
+ - ASTERISK-30186: res_pjsip: Add support for reloading TLS certificate and key information
+ - ASTERISK-30190: res_geolocation: GEOLOC_PROFILE isn't returning correct values on incoming channel
+ - ASTERISK-30192: res_tonedetect: fix typo for frametype
+ - ASTERISK-30193: chan_pjsip should return all codecs on a re-INVITE without SDP
+ - ASTERISK-30198: Error `Too many open files` occurs after about ~8000 calls when using mixmonitor
+ - ASTERISK-30209: pbx_variables: Use const char for pbx_substitute_variables_helper_full_location
+ - ASTERISK-30210: func_frame_trace: Channel masquerade triggers assertion
+ - ASTERISK-30211: app_confbridge: Add end_marked_any option
+ - ASTERISK-30213: Make crypto_load() reentrant and handle symlinks correctly
+ - ASTERISK-30215: Inbound SIP INVITE with Geo Location causing a Segmentation Fault
+ - ASTERISK-30216: app_bridgewait: Add option for BridgeWait to not answer
+ - ASTERISK-30217: Registration do not allow multiple proxies
+ - ASTERISK-30220: func_scramble: Fix segfault due to null pointer deref
+ - ASTERISK-30222: func_strings: Add trim functions
+ - ASTERISK-30223: features: add no-answer option to Bridge application
+ - ASTERISK-30226: REGRESSION: res_crypto complains about the stir_shaken directory in /var/lib/asterisk/keys
+ - ASTERISK-30232: Initialize stack-based ast_test_capture structures correctly
+ - ASTERISK-30234: res_geolocation: ...may be used uninitialized error in geoloc_config.c
+ - ASTERISK-30235: res_crypto and tests: Memory issues and and uninitialized variable error
+ - ASTERISK-30237: res_prometheus: Crash when scraping bridges
+ - ASTERISK-30239: Prometheus plugin crashes Asterisk when using local channel
+ - ASTERISK-30240: app voicemail odbc build error with gcc 11.1
+ - ASTERISK-30241: res_pjsip_gelocation: Downgrade some NOTICE scope trace debugs to DEBUG level
+ - ASTERISK-30243: func_logic: IF function complains if both branches are empty
+ - ASTERISK-30244: res_pjsip_pubsub: Occasional crash when TCP/TLS connection terminated and subscription persistence is removed
+ - ASTERISK-30245: db: ListItems is incorrect
+ - ASTERISK-30248: ast_get_digit_str adds bogus initial delimiter if first character not to be spoken
+ - ASTERISK-30252: Unidirectional snoop on resampled channel causes garbled audio
+ - ASTERISK-30254: res_tonedetect: Add audible ringback detection to TONE_DETECT
+ - ASTERISK-30256: chan_dahdi: Fix format truncation warnings
+ - ASTERISK-30258: Dialing API: Cancel a running async thread, does not always cancel all calls
+ - ASTERISK-30262: res_pjsip_session: Allow a context to be specified for overlap dialing
+ - ASTERISK-30263: res_pjsip_notify: Allow using pjsip_notify.conf from AMI
+ - ASTERISK-30264: res_pjsip: Subscription handlers do not get cleanly unregistered, causing crash
+ - ASTERISK-30265: res_pjsip_session: Fix missing PLAR support on INVITEs
+ - ASTERISK-30273: test_mwi: compilation fails on 32-bit Debian
+ - ASTERISK-30274: chan_dahdi: Unavailable channels are BUSY
+ - ASTERISK-30278: tcptls: Abort occurs if SSL error is logged if MALLOC_DEBUG is enabled
+ - ASTERISK-30280: Create capability to assign a Media Experience Score to RTP streams
+ - ASTERISK-30281: chan_rtp: Local address being used before being set
+ - ASTERISK-30282: CI: Coredump output isn't saved when running unittests
+ - ASTERISK-30283: app_voicemail: Fix msg_create_from_file not sending email to user
+ - ASTERISK-30284: app_mixmonitor: Add option to delete recording file when done
+ - ASTERISK-30285: manager.c: Remove outdated documentation
+ - ASTERISK-30286: app_mixmonitor: Add option to use real Caller ID for Caller ID
+ - ASTERISK-30289: xmldoc: Allow XML docs to be reloaded
+ - ASTERISK-30290: file.c: Don't emit warnings on winks.
+ - ASTERISK-30293: Memory leak in JSON_DECODE
+ - ASTERISK-30295: test_json: Remove duplicated static function
+ - ASTERISK-30305: chan_dahdi: Allow FXO channels to start immediately
+ - ASTERISK-30308: pbx_builtins: Allow Answer to return immediately
+ - ASTERISK-30311: func_presencestate: Fix invalid memory access.
+ - ASTERISK-30314: res_agi: RECORD FILE doesn't respect "transmit_silence" asterisk.conf option
+ - ASTERISK-30316: res_pjsip: Documentation should point out default if contact_user is not being set for outbound registrations
+ - ASTERISK-30319: Add BYE Reason support for SIP
+ - ASTERISK-30321: Build: Embedded blobs have executable stacks
+ - ASTERISK-30322: res_hep: Add capture agent name support
+ - ASTERISK-30325: Upgrade Asterisk to bundled pjproject 2.13
+ - ASTERISK-30326: app_followme: Setting enable_callee_prompt=no(false) in followme.conf breaks timeout for calling from FollowMe application
+ - ASTERISK-30327: rtp_engine.h: Remove obsolete example usage
+ - ASTERISK-30328: Typo in from_domain description on res_pjsip configuration documentation
+ - ASTERISK-30330: callerid: Allow timezone to be specified at runtime
+ - ASTERISK-30331: sig_analog: Add full Caller ID support for incoming calls
+ - ASTERISK-30332: func_callerid: Warn if invalid redirecting reason provided
+ - ASTERISK-30333: chan_dahdi: Fix broken presentation for FXO caller ID
+ - ASTERISK-30336: sig_analog: Fix no timeout duration
+ - ASTERISK-30338: pjproject: Backport security fixes from 2.13
+ - ASTERISK-30340: res_media_cache curl options configureable
+ - ASTERISK-30344: ari: Memory leak in create when specifying JSON
+ - ASTERISK-30345: loader.c: Modules that decline to load cannot be reloaded
+ - ASTERISK-30346: Fix NULL dereferencing issue in Geolocation
+ - ASTERISK-30347: xmldocs: Remove references to removed applications
+ - ASTERISK-30349: app_if: Format truncation error
+ - ASTERISK-30350: res_pjsip_sdp_rtp: rtp_timeout_hold is not used when moh_passthrough has call on hold
+ - ASTERISK-30351: manager: Originate variables are not added when setvar used in manager.conf
+ - ASTERISK-30353: func_frame_trace: Print text for text frames
+ - ASTERISK-30354: chan_iax2: Lack of formats prior to receiving voice frames causes jitterbuffer to stall
+ - ASTERISK-30357: chan_dahdi: Allow automatic reoriginate on hangup
+ - ASTERISK-30359: Install Prereq Script Enhancements
+ - ASTERISK-30361: json.h: Add missing ast_json_object_real_get
+ - ASTERISK-30367: pbx: Fix outdated channel snapshots with pbx_exec
+ - ASTERISK-30369: res_pjsip: Websockets from same IP shut down when they shouldn't be
+ - ASTERISK-30372: sig_analog: Add Called Subscriber Held capability
+ - ASTERISK-30375: res_http_media_cache: Crash when URL has no path component.
+ - ASTERISK-30379: http: fix NULL pointer dereference while enable_status on TLS-only
+ - ASTERISK-30388: res_phoneprov: Stale SERVER variable when multi-homed
+ - ASTERISK-30391: res_rtp_asterisk: Issue with transcoding g722 after MES changes
+ - ASTERISK-30404: app_directory: Add reading directory configuration from custom file
+ - ASTERISK-30405: app_directory: Add 's' option to skip channel call
+ - ASTERISK-30406: pbx_ael: Global variables are not expanded.
+ - ASTERISK-30407: res_stir_shaken: Ordering of JSON fields incorrect, and tn lacks canonicalization
+ - ASTERISK-30411: app_read: add option to include terminating digit on empty, terminated strings
+ - ASTERISK-30417: Copy/Paste error in UnpauseQueueMember
+ - ASTERISK-30419: pjsip: Crash when sending NOTIFY in PJSIP 2.13
+ - ASTERISK-30422: app_senddtmf: add the option for senddtmf to answer
+ - ASTERISK-30424: pjproject_bundled: cross-compilation broken when ssl autodetected
+ - ASTERISK-30428: bridging: Music on hold continues after INVITE with replaces
+ - ASTERISK-30429: res_sorcery_memory_cache: Memory leak when calling sorcery_memory_cache_open
+ - ASTERISK-30437: app_queue ability to start periodic announcements at a different time than the playing interval (frequency)
+ - ASTERISK-30440: app_senddtmf: Add Flash AMI action
+ - ASTERISK-30441: func_json: Fix JSON parsing of complex objects
+ - ASTERISK-30442: make install-logrotate causes logrotate to fail on service restart
+ - ASTERISK-30446: bridge_builtin_features: add periodic beep option to one touch monitor
+ - ASTERISK-30449: contrib: rc.archlinux.asterisk uses invalid redirect.
+ - ASTERISK-30455: Increase channel name column width on cli
+ - ASTERISK-30457: res_agi: RECORD FILE plays 2 beeps
+ - ASTERISK-30462: res_musiconhold: Add looplast option
+ - ASTERISK-30464: app_mixmonitor: Allow specifying which MixMonitor instance (or all of them) to mute/unmute using MixMonitorMute
+ - ASTERISK-30465: format_sln: add support for .slin files
+ - ASTERISK-30469: res_pjsip_pubsub: Regression for subscription shutdowns
+ - ASTERISK-30472: pbx_ael: Literal usage for variables broken
+ - ASTERISK-30474: res_prometheus provides broken description
+ - ASTERISK-30479: voicemail.conf: Comments about #include files are wrong
+ - ASTERISK-30483: logger: Allow filtering logs in CLI by channel
+ - ASTERISK-30486: app_queue: Fix minor xmldoc issues
+
+### Commits By Author:
+
+- ### Alexander Greiner-Baer (1):
+ - res_pjsip: set Accept-Encoding to identity in OPTIONS response
+
+- ### Alexander Traud (67):
+ - samples: Fix keep_alive_interval default in pjsip.conf.
+ - sip_nat_settings: Update script for latest Linux.
+ - BuildSystem: Enable Lua 5.4.
+ - install_prereq: Add GMime 3.0.
+ - chan_sip: On authentication, pick MD5 for sure.
+ - Compiler fixes for GCC with -Os
+ - Compiler fixes for GCC when printf %s is NULL
+ - Compiler fixes for GCC with -Og
+ - res_stir_shaken: Include OpenSSL headers where used actually.
+ - res_pjsip/config_transport: Load and run without OpenSSL.
+ - modules.conf: Align the comments for more conclusiveness.
+ - chan_sip: Remove unused sip_socket->port.
+ - loader: Sync load- and build-time deps.
+ - codecs: Remove test-law.
+ - chan_sip: SDP: Sidestep stream parsing when its media is disabled.
+ - res_pjsip_session: Avoid sometimes-uninitialized warning with Clang.
+ - pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.
+ - chan_sip: SDP: Reject audio streams correctly.
+ - channel: Set up calls without audio (text+video), again.
+ - chan_sip: Set up calls without audio (text+video), again.
+ - chan_sip: Allow [peer] without audio (text+video).
+ - rtp: Enable srtp replay protection
+ - chan_sip: Filter pass-through audio/video formats away, again.
+ - res_format_attr_h263: Generate valid SDP fmtp for H.263+.
+ - chan_iax2: System Header strings is included via asterisk.h/compat.h.
+ - res_format_attr_*: Parameter Names are Case-Insensitive.
+ - aelparse: Accept an included context with timings.
+ - BuildSystem: Remove two dead exceptions for compiler Clang.
+ - dialplan: Add one static and fix two whitespace errors.
+ - res_rtp_asterisk: sqrt(.) requires the header math.h.
+ - stasis: Avoid 'dispatched' as unused variable in normal mode.
+ - res_config_sqlite: Remove deprecated module.
+ - res_snmp: As build tool, prefer pkg-config over net-snmp-config.
+ - BuildSystem: In POSIX sh, == in place of = is undefined.
+ - progdocs: Avoid 'name' with Doxygen \file.
+ - bridge_channel: Fix for Doxygen.
+ - progdocs: Use Doxygen \example correctly.
+ - progdocs: Avoid multiple use of section labels.
+ - tests: Fix for Doxygen.
+ - apps: Fix for Doxygen.
+ - addons: Fix for Doxygen.
+ - bridges: Fix for Doxygen.
+ - res_pjsip: Fix for Doxygen.
+ - chan_iax2: Fix for Doxygen.
+ - channel: Fix for Doxygen.
+ - res_xmpp: Fix for Doxygen.
+ - app: Fix for Doxygen.
+ - stasis: Fix for Doxygen.
+ - BuildSystem: Consistently allow 'ye' even for Jansson.
+ - ari-stubs: Avoid 'is' as comparism with an literal.
+ - frame: Fix for Doxygen.
+ - res_ari: Fix for Doxygen.
+ - parking: Fix for Doxygen.
+ - odbc: Fix for Doxygen.
+ - channels: Fix for Doxygen.
+ - xmldoc: Fix for Doxygen.
+ - progdocs: Remove outdated references in doxyref.h.
+ - stir/shaken: Avoid a compiler extension of GCC.
+ - progdocs: Fix grouping for latest Doxygen.
+ - progdocs: Fix for Doxygen, the hidden parts.
+ - main: Fix for Doxygen.
+ - res: Fix for Doxygen.
+ - res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites.
+ - progdocs: Update Makefile.
+ - xmldoc: Correct definition for XML element 'matchInfo'.
+ - progdocs: Fix Doxygen left-overs.
+ - xmldoc: Avoid whitespace around value for parameter/required.
+
+- ### Alexandre Fournier (1):
+ - res_geoloc: fix NULL pointer dereference bug
+
+- ### Alexei Gradinari (12):
+ - sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data
+ - res_fax: validate the remote/local Station ID for UTF-8 format
+ - app_queue: load queues and members from Realtime when needed
+ - res_pjsip_pubsub: provide a display name for RLS subscriptions
+ - res_pjsip_pubsub: fix Batched Notifications stop working
+ - res_pjsip_pubsub: update RLS to reflect the changes to the lists
+ - res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request
+ - res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI
+ - res_pjsip_pubsub: XML sanitized RLS display name
+ - res_pjsip_pubsub: delete scheduled notification on RLS update
+ - res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update
+ - format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
+
+- ### Andre Barbosa (3):
+ - res_stasis_playback: Send PlaybackFinish event only once for errors
+ - res_stasis_playback: Check for chan hangup on play_on_channels
+ - media_cache: Don't lock when curl the remote file
+
+- ### Andrew Siplas (1):
+ - logger.conf.sample: add missing comment mark
+
+- ### Bastian Triller (2):
+ - res_pjsip_session: Send Session Interval too small response
+ - func_json: Fix crashes for some types
+
+- ### Ben Ford (27):
+ - res_stir_shaken: Fix memory allocation error in curl.c
+ - utils.c: NULL terminate ast_base64decode_string.
+ - Bridging: Use a ref to bridge_channel's channel to prevent crash.
+ - AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
+ - chan_pjsip.c: Add parameters to frame in indicate.
+ - core_unreal: Fix T.38 faxing when using local channels.
+ - res_pjsip_session.c: Check topology on re-invite.
+ - AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
+ - logging: Add .log to samples and update asterisk.logrotate.
+ - logger.conf.sample: Add more debug documentation.
+ - res_aeap: Add basic config skeleton and CLI commands.
+ - STIR/SHAKEN: Fix certificate type and storage.
+ - STIR/SHAKEN: OPENSSL_free serial hex from openssl.
+ - STIR/SHAKEN: Switch to base64 URL encoding.
+ - STIR/SHAKEN: Add Date header, dest->tn, and URL checking.
+ - Update AMI and ARI versions for Asterisk 20.
+ - STIR/SHAKEN: Option split and response codes.
+ - AST-2022-001 - res_stir_shaken/curl: Limit file size and check start.
+ - AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header.
+ - res_pjsip_stir_shaken.c: Fix enabled when not configured.
+ - res_pjsip: Add TEL URI support for basic calls.
+ - pjproject: 2.13 security fixes
+ - res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
+ - AMI: Add CoreShowChannelMap action.
+ - res_pjsip_session: Added new function calls to avoid ABI issues.
+ - manager.c: Prevent path traversal with GetConfig.
+ - Upgrade bundled pjproject to 2.14.
+
+- ### Bernd Zobl (2):
+ - res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter
+ - res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress
+
+- ### Boris P. Korzun (10):
+ - bridge_basic: Fixed setup of recall channels
+ - res_musiconhold: Add support of various URL-schemes by MoH.
+ - format_wav: Support of MIME-type for wav16
+ - res_config_pgsql: Limit realtime_pgsql() to return one (no more) record.
+ - rtp_engine: Add type field for JSON RTCP Report stasis messages
+ - res_config_pgsql: Add text-type column check in require_pgsql()
+ - res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity
+ - res_prometheus: Optional load res_pjsip_outbound_registration.so
+ - pbx_lua: Remove compiler warnings
+ - http.c: Fix NULL pointer dereference bug
+
+- ### Brad Smith (4):
+ - res_rtp_asterisk.c: Fix runtime issue with LibreSSL
+ - main/utils: Implement ast_get_tid() for OpenBSD
+ - main/utils: Simplify the FreeBSD ast_get_tid() handling
+ - BuildSystem: Bump autotools versions on OpenBSD.
+
+- ### Carlos Oliva (1):
+ - app_mp3: Force output to 16 bits in mpg123
+
+- ### Christof Efkemann (1):
+ - app_sayunixtime: Use correct inflection for German time.
+
+- ### Dan Cropp (2):
+ - chan_pjsip: Incorporate channel reference count into transfer_refer().
+ - chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable
+
+- ### Dennis Buteyn (1):
+ - chan_sip: Clear ToHost property on peer when changing to dynamic host
+
+- ### Dovid Bender (1):
+ - func_curl.c: Allow user to set what return codes constitute a failure.
+
+- ### Dustin Marquess (1):
+ - res_fax_spandsp: Add spandsp 3.0.0+ compatibility
+
+- ### Eduardo (1):
+ - codec_builtin: Use multiples of 20 for maximum_ms
+
+- ### Evandro César Arruda (1):
+ - app_queue: Member lastpause time reseting
+
+- ### Evgenios_Greek (1):
+ - stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing
+
+- ### Fabrice Fontaine (2):
+ - main/iostream.c: fix build with libressl
+ - configure: fix detection of re-entrant resolver functions
+
+- ### Florentin Mayer (1):
+ - res_pjsip_sdp_rtp: Preserve order of RTP codecs
+
+- ### Frederic LE FOLL (1):
+ - Dialing API: Cancel a running async thread, may not cancel all calls
+
+- ### Frederic Van Espen (1):
+ - ast_coredumper: Fix deleting results when output dir is set
+
+- ### George Joseph (139):
+ - Prepare master for the next Asterisk version
+ - CI: Force publishAsteriskDocs to use python2
+ - res_pjsip_session: Ensure reused streams have correct bundle group
+ - ACN: Configuration renaming for pjsip endpoint
+ - ACN: Changes specific to the core
+ - stream.c: Added 2 more debugging utils and added pos to stream string
+ - scope_trace: Added debug messages and added additional macros
+ - logger.c: Added a new log formatter called "plain"
+ - ast_coredumper: Fix issues with naming
+ - res_pjsip_session: Handle multi-stream re-invites better
+ - debugging: Add enough to choke a mule
+ - res_pjsip_session: Fix issue with COLP and 491
+ - bridge_softmix/sfu_topologies_on_join: Ignore topology change failures
+ - logger.h: Fix ast_trace to respect scope_level
+ - app_confbridge/bridge_softmix: Add ability to force estimated bitrate
+ - pjsip_scheduler.c: Add type ONESHOT and enhance cli show command
+ - res_pjsip_outbound_registration.c: Use our own scheduler and other stuff
+ - app_queue: Fix deadlock between update and show queues
+ - logger.c: Automatically add a newline to formats that don't have one
+ - Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other st..
+ - chan_iax2.c: Require secret and auth method if encryption is enabled
+ - res_pjsip_refer: Always serialize calls to refer_progress_notify
+ - res_pjsip_refer: Refactor progress locking and serialization
+ - res_pjsip_refer: Move the progress dlg release to a serializer
+ - res_pjsip_session: Make reschedule_reinvite check for NULL topologies
+ - res_prometheus: Clone containers before iterating
+ - bridge_channel_write_frame: Check for NULL channel
+ - res_pjsip: Update documentation for the auth object
+ - res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs
+ - res_pjsip_messaging: Refactor outgoing URI processing
+ - res_pjsip_messaging: Overwrite user in existing contact URI
+ - jitterbuffer: Correct signed/unsigned mismatch causing assert
+ - res_pjproject: Allow mapping to Asterisk TRACE level
+ - bridge_softmix: Suppress error on topology change failure
+ - res_snmp: Add -fPIC to _ASTCFLAGS
+ - pjproject: Add patch to fix trailing whitespace issue in rtpmap
+ - BuildSystem: Check for alternate openssl packages
+ - ast_coredumper: Refactor to better find things
+ - CI: Rename 'master' node to 'built-in'
+ - bundled_pjproject: Add more support for multipart bodies
+ - bundled_pjproject: Make it easier to hack
+ - bundled_pjproject: Create generic pjsip_hdr_find functions
+ - res_pjsip: Add utils for checking media types
+ - build: Fix issues building pjproject
+ - res_pjsip: Make message_filter and session multipart aware
+ - bundled_pjproject: Fix srtp detection
+ - res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup
+ - build: Add "basebranch" to .gitreview
+ - bundled_pjproject: Add additional multipart search utils
+ - build: Refactor the earlier "basebranch" commit
+ - Makefile: Allow XML documentation to exist outside source files
+ - core: Config and XML tweaks needed for geolocation
+ - xmldoc: Fix issue with xmlstarlet validation
+ - xml.c, config,c: Add stylesheets and variable list string parsing
+ - make_xml_documentation: Remove usage of get_sourceable_makeopts
+ - Makefile: Disable XML doc validation
+ - GCC12: Fixes for 18+. state_id_by_topic comparing wrong value
+ - GCC12: Fixes for 16+
+ - Geolocation: Base Asterisk Prereqs
+ - Geolocation: Core Capability Preview
+ - Geolocation: chan_pjsip Capability Preview
+ - geoloc_eprofile.c: Fix setting of loc_src in set_loc_src()
+ - Update defaultbranch to 20
+ - Geolocation: Wiki Documentation
+ - res_geolocation: Address user issues, remove complexity, plug leaks
+ - res_geolocation: Add built-in profiles
+ - res_geolocation: Add profile parameter suppress_empty_ca_elements
+ - res_geolocation: Allow location parameters on the profile object
+ - res_geolocation: Add two new options to GEOLOC_PROFILE
+ - res_geolocation: Fix segfault when there's an empty element
+ - res_geolocation: Fix issues exposed by compiling with -O2
+ - res_crypto: Memory issues and uninitialized variable errors
+ - res_geolocation: Update wiki documentation
+ - chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer
+ - runUnittests.sh: Save coredumps to proper directory
+ - pjsip_transport_events: Fix possible use after free on transport
+ - res_rtp_asterisk: Asterisk Media Experience Score (MES)
+ - res_pjsip_transport_websocket: Add remote port to transport
+ - Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"
+ - res_rtp_asterisk: Asterisk Media Experience Score (MES)
+ - res_rtp_asterisk: Don't use double math to generate timestamps
+ - res_pjsip: Replace invalid UTF-8 sequences in callerid name
+ - make_version: Strip svn stuff and suppress ref HEAD errors
+ - test.c: Fix counting of tests and add 2 new tests
+ - Initial GitHub Issue Templates
+ - Initial GitHub PRs
+ - Set up new ChangeLogs directory
+ - apply_patches: Sort patch list before applying
+ - build: Fix a few gcc 13 issues
+ - test_stasis_endpoints.c: Make channel_messages more stable
+ - test_statis_endpoints: Fix channel_messages test again
+ - rest-api: Updates for new documentation site
+ - doc: Remove obsolete CHANGES-staging and UPGRADE-staging
+ - app.h: Move declaration of ast_getdata_result before its first use
+ - pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+ - rest-api: Run make ari-stubs
+ - download_externals: Fix a few version related issues
+ - alembic: Fix quoting of the 100rel column
+ - ari-stubs: Fix broken documentation anchors
+ - ari-stubs: Fix more local anchor references
+ - ari-stubs: Fix more local anchor references
+ - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+ - res_rtp_asterisk: Fix regression issues with DTLS client check
+ - Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
+ - safe_asterisk: Change directory permissions to 755
+ - func_periodic_hook: Don't truncate channel name
+ - make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
+ - file.c: Add ability to search custom dir for sounds
+ - asterisk.c: Use the euid's home directory to read/write cli history
+ - lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
+ - Add libjwt to third-party
+ - logger.h: Add ability to change the prefix on SCOPE_TRACE output
+ - res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
+ - api.wiki.mustache: Fix indentation in generated markdown
+ - bridge_simple: Suppress unchanged topology change requests
+ - chan_pjsip: Add PJSIPHangup dialplan app and manager action
+ - codec_ilbc: Disable system ilbc if version >= 3.0.0
+ - SECURITY.md: Update with correct documentation URL
+ - ast_coredumper: Increase reliability
+ - MergeApproved.yml: Remove unneeded concurrency
+ - Revert "core & res_pjsip: Improve topology change handling."
+ - Reduce startup/shutdown verbose logging
+ - pjsip show channelstats: Prevent possible segfault when faxing
+ - Stir/Shaken Refactor
+ - Makefile: Add stir_shaken/cache to directories created on install
+ - attestation_config.c: Use ast_free instead of ast_std_free
+ - res_pjsip_stir_shaken.c: Add checks for missing parameters
+ - Initial commit for certified-20.7
+ - Fix incorrect application and function documentation references
+ - manager.c: Add missing parameters to Login documentation
+ - rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
+ - make_buildopts_h: Always include DETECT_DEADLOCKS
+ - tcptls/iostream: Add support for setting SNI on client TLS connections
+ - res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2)
+ - logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
+ - stir_shaken: Fix memory leak, typo in config, tn canonicalization
+ - stasis_channels: Use uniqueid and name to delete old snapshots
+ - security_agreement.c: Always add the Require and Proxy-Require headers
+ - app_voicemail_odbc: Allow audio to be kept on disk
+
+- ### Gitea (1):
+ - res_pjsip_header_funcs: Duplicate new header value, don't copy.
+
+- ### Guido Falsi (1):
+ - res_rtp_asterisk.c: Fix build failure when not building with pjproject.
+
+- ### Henning Westerholt (3):
+ - res_pjsip: return all codecs on a re-INVITE without SDP
+ - chan_pjsip: fix music on hold continues after INVITE with replaces
+ - chan_pjsip: also return all codecs on empty re-INVITE for late offers
+
+- ### Holger Hans Peter Freyther (9):
+ - res_pjsip_sdp_rtp: Fix accidentally native bridging calls
+ - pjsip: Generate progress (once) when receiving a 180 with a SDP
+ - res_prometheus: Do not crash on invisible bridges
+ - res_http_media_cache: Do not crash when there is no extension
+ - res_http_media_cache: Introduce options and customize
+ - res_prometheus: Do not generate broken metrics
+ - ari/stasis: Indicate progress before playback on a bridge
+ - ari: Provide the caller ID RDNIS for the channels
+ - stasis: Update the snapshot after setting the redirect
+
+- ### Hugh McMaster (1):
+ - configure.ac: Use pkg-config to detect libxml2
+
+- ### Igor Goncharovsky (5):
+ - res_ari: Fix audiosocket segfault
+ - res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern
+ - res_pjsip_outbound_registration: Allow to use multiple proxies for registration
+ - res_pjsip: Fix path usage in case dialing with '@'
+ - res_pjsip_rfc3326: Add SIP causes support for RFC3326
+
+- ### Ivan Poddubny (1):
+ - asterisk.c: Fix sending incorrect messages to systemd notify
+
+- ### Ivan Poddubnyi (5):
+ - chan_pjsip: Stop queueing control frames twice on outgoing channels
+ - chan_pjsip: Assign SIPDOMAIN after creating a channel
+ - main/frame: Add missing control frame names to ast_frame_subclass2str
+ - res_pjsip_diversion: Fix adding more than one histinfo to Supported
+ - app_queue: Fix conversion of complex extension states into device states
+
+- ### Jaco Kroon (22):
+ - pbx_lua: Add LUA_VERSIONS environment variable to ./configure.
+ - func_lock: fix multiple-channel-grant problems.
+ - contrib/systemd: Added note on common issues with systemd and asterisk
+ - AC_HEADER_STDC causes a compile failure with autoconf 2.70
+ - func_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
+ - app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS
+ - res_odbc_transaction: correctly initialise forcecommit value from DSN.
+ - app.h: Fix -Werror=zero-length-bounds compile errors in dev mode.
+ - func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds
+ - menuselect: exit non-zero in case of failure on --enable|disable options.
+ - func_lock: Fix requesters counter in error paths.
+ - func_lock: Fix memory corruption during unload.
+ - func_lock: Prevent module unloading in-use module.
+ - func_lock: Add "dialplan locks show" cli command.
+ - logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__
+ - manager: be more aggressive about purging http sessions.
+ - Build system: Avoid executable stack.
+ - app_queue: periodic announcement configurable start time.
+ - res_calendar: output busy state as part of show calendar.
+ - configure: fix test code to match gethostbyname_r prototype.
+ - tcptls: when disabling a server port, we should set the accept_fd to -1.
+ - app_queue: periodic announcement configurable start time.
+
+- ### Jason D. McCormick (1):
+ - install_prereq: Fix dependency install on aarch64.
+
+- ### Jasper Hafkenscheid (1):
+ - res_srtp: Disable parsing of not enabled cryptos
+
+- ### Jasper van der Neut (1):
+ - channels: Don't dereference NULL pointer
+
+- ### Jean Aunis (4):
+ - resource_endpoints.c: memory leak when providing a 404 response
+ - Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
+ - translate.c: Take sampling rate into account when checking codec's buffer size
+ - res_rtp_asterisk: fix memory leak
+
+- ### Jeremy Lainé (1):
+ - res_rtp_asterisk: make it possible to remove SOFTWARE attribute
+
+- ### Jiajian Zhou (1):
+ - AMI: Add parking position parameter to Park action
+
+- ### Joe Searle (1):
+ - res_stasis.c: Add new type 'sdp_label' for bridge creation.
+
+- ### Jose Lopes (1):
+ - res_pjsip_header_funcs: Add functions PJSIP_RESPONSE_HEADER and PJSIP_RESPONSE..
+
+- ### Joseph Nadiv (3):
+ - res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml
+ - res_pjsip.c: Support endpoints with domain info in username
+ - res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts
+
+- ### Josh Soref (25):
+ - agi: Spelling fixes
+ - rest-api-templates: Spelling fixes
+ - res: Spelling fixes
+ - Makefile: Spelling fixes
+ - utils: Spelling fixes
+ - main: Spelling fixes
+ - pbx: Spelling fixes
+ - funcs: Spelling fixes
+ - CHANGES: Spelling fixes
+ - tests: Spelling fixes
+ - channels: Spelling fixes
+ - apps: Spelling fixes
+ - bridges: Spelling fixes
+ - UPGRADE.txt: Spelling fixes
+ - include: Spelling fixes
+ - menuselect: Spelling fixes
+ - doc: Spelling fixes
+ - configs: Spelling fixes
+ - addons: Spelling fixes
+ - CREDITS: Spelling fixes
+ - formats: Spelling fixes
+ - codecs: Spelling fixes
+ - contrib: Spelling fixes
+ - build_tools: Spelling fixes
+ - test_time.c: Tolerate DST transitions
+
+- ### Joshua C. Colp (85):
+ - pjsip: Include timer patch to prevent cancelling timer 0.
+ - websocket / pjsip: Increase maximum packet size.
+ - res_pjsip_registrar: Don't specify an expiration for static contacts.
+ - res_pjsip: Fix codec preference defaults.
+ - res_pjsip_session: Don't aggressively terminate on failed re-INVITE.
+ - pbx: Fix hints deadlock between reload and ExtensionState.
+ - parking: Copy parker UUID as well.
+ - res_pjsip_session: Fix session reference leak.
+ - res_pjsip_session: Fix stream name memory leak.
+ - res_pjsip: Adjust outgoing offer call pref.
+ - voicemail: add option 'e' to play greetings as early media
+ - pjsip: Match lifetime of INVITE session to our session.
+ - res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
+ - pjsip: Make modify_local_offer2 tolerate previous failed SDP.
+ - res_pjsip_session: Always produce offer on re-INVITE without SDP.
+ - channel: Fix memory leak in suppress API.
+ - res_pjsip_nat: Don't rewrite Contact on REGISTER responses.
+ - asterisk: Update copyright.
+ - res_pjsip_registrar: Include source IP and port in log messages.
+ - sorcery: Add support for more intelligent reloading.
+ - channel: Fix crash in suppress API.
+ - documentation: Fix non-matching module support levels.
+ - xml: Embed module information into core XML documentation.
+ - xml: Allow deprecated_in and removed_in for MODULEINFO.
+ - menuselect: Add ability to set deprecated and removed versions.
+ - res_rtp_asterisk: Force resync on SSRC change.
+ - res_pjsip: Add support for partial transport reload.
+ - core_unreal: Fix deadlock with T.38 control frames.
+ - app_queue: Only send QueueMemberStatus if status changes.
+ - res_pjsip: Give error when TLS transport configured but not supported.
+ - res_rtp_asterisk: Only raise flash control frame on end.
+ - loader: Output warnings for deprecated modules.
+ - svn: Switch to https scheme.
+ - chan_local: Skip filtering audio formats on removed streams.
+ - pjsip: Add patch for resolving STUN packet lifetime issues.
+ - asterisk: We've moved to Libera Chat!
+ - res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.
+ - res_pjsip: On partial transport reload also move factories.
+ - core: Don't play silence for Busy() and Congestion() applications.
+ - AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.
+ - docs: Remove embedded macro in WaitForCond XML documentation.
+ - policy: Deprecate modules and add versions to others.
+ - cdr_mysql: Remove deprecated module.
+ - app_mysql: Remove deprecated module.
+ - app_ices: Remove deprecated module.
+ - app_fax: Remove deprecated module.
+ - app_url: Remove deprecated module.
+ - app_image: Remove deprecated module.
+ - app_nbscat: Remove deprecated module.
+ - app_dahdiras: Remove deprecated module.
+ - cdr_syslog: Remove deprecated module.
+ - chan_oss: Remove deprecated module.
+ - chan_phone: Remove deprecated module.
+ - chan_nbs: Remove deprecated module.
+ - chan_misdn: Remove deprecated module.
+ - chan_vpb: Remove deprecated module.
+ - res_config_sqlite: Remove deprecated module.
+ - conf2ael: Remove deprecated application.
+ - muted: Remove deprecated application.
+ - res_monitor: Disable building by default.
+ - ari: Ignore invisible bridges when listing bridges.
+ - bridge: Deny full Local channel pair in bridge.
+ - bridge: Unlock channel during Local peer check.
+ - jansson: Update bundled to 2.14 version.
+ - pjproject: Update bundled to 2.12 release.
+ - func_odbc: Add SQL_ESC_BACKSLASHES dialplan function.
+ - pjsip: Increase maximum number of format attributes.
+ - cdr_adaptive_odbc: Add support for SQL_DATETIME field type.
+ - res_pjsip: Always set async_operations to 1.
+ - manager: Terminate session on write error.
+ - res_pjsip_transport_websocket: Also set the remote name.
+ - websocket / aeap: Handle poll() interruptions better.
+ - pjsip: Add TLS transport reload support for certificate and key.
+ - res_pjsip_sdp_rtp: Skip formats without SDP details.
+ - res_agi: Respect "transmit_silence" option for "RECORD FILE".
+ - ari: Destroy body variables in channel create.
+ - res_pjsip_aoc: Don't assume a body exists on responses.
+ - pbx_dundi: Fix PJSIP endpoint configuration check.
+ - LICENSE: Update link to trademark policy.
+ - app_queue: Add support for applying caller priority change immediately.
+ - audiohook: Unlock channel in mute if no audiohooks present.
+ - manager: Tolerate stasis messages with no channel snapshot.
+ - variables: Add additional variable dialplan functions.
+ - Update config.yml
+ - utils: Make behavior of ast_strsep* match strsep.
+
+- ### Joshua Colp (1):
+ - Revert "app_queue: periodic announcement configurable start time."
+
+- ### Kevin Harwell (28):
+ - chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution
+ - conversions: Add string to signed integer conversion functions
+ - Logging: Add debug logging categories
+ - res_pjsip, res_pjsip_session: initialize local variables
+ - AST-2020-001 - res_pjsip: Return dialog locked and referenced
+ - pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type
+ - app_mixmonitor: cleanup datastore when monitor thread fails to launch
+ - AST-2021-002: Remote crash possible when negotiating T.38
+ - res_rtp_asterisk: Add packet subtype during RTCP debug when relevant
+ - time: Add timeval create and unit conversion functions
+ - res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
+ - res_rtp_asterisk: Statically declare rtp_drop_packets_data object
+ - res_rtp_asterisk: Don't count 0 as a minimum lost packets
+ - res_rtp_asterisk: Fix standard deviation calculation
+ - AST-2021-008 - chan_iax2: remote crash on unsupported media format
+ - AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
+ - format_ogg_speex: Implement a "not supported" write handler
+ - res_speech: Add a type conversion, and new engine unregister methods
+ - strings/json: Add string delimter match, and object create with vars methods
+ - http.c: Add ability to create multiple HTTP servers
+ - tcptls.c: refactor client connection to be more robust
+ - res_http_websocket: Add a client connection timeout
+ - deprecation cleanup: remove leftover files
+ - res_pjsip_header_funcs: wrong pool used tdata headers
+ - res_aeap & res_speech_aeap: Add Asterisk External Application Protocol
+ - test_aeap_transport: disable part of failing unit test
+ - res_pjsip: allow TLS verification of wildcard cert-bearing servers
+ - cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type
+
+- ### Kfir Itzhak (2):
+ - app_queue: Fix leave-empty not recording a call as abandoned
+ - app_queue: Add QueueWithdrawCaller AMI action
+
+- ### Luke Escude (1):
+ - res_pjsip_sdp_rtp.c: Support keepalive for video streams.
+
+- ### Marcel Wagner (3):
+ - documentation: Add information on running install_prereq script in readme
+ - res_pjsip: Update contact_user to point out default
+ - res_pjsip: Fix typo in from_domain documentation
+
+- ### Mark Murawski (4):
+ - logger: Console sessions will now respect logger.conf dateformat= option
+ - pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+ - pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+ - Remove files that are no longer updated
+
+- ### Mark Petersen (10):
+ - apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL..
+ - app_voicemail.c: Support for Danish syntax in VM
+ - app_queue.c: added DIALEDPEERNUMBER on outgoing channel
+ - app_queue.c: Support for Nordic syntax in announcements
+ - app_queue.c: Queue don't play "thank-you" when here is no hold time announceme..
+ - chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN
+ - res_prometheus.c: missing module dependency
+ - chan_sip: SIP route header is missing on UPDATE
+ - chan_pjsip: add allow_sending_180_after_183 option
+ - chan_sip.c Session timers get removed on UPDATE
+
+- ### Martin Nystroem (1):
+ - res_ari.c: Add additional output to ARI requests when debug is enabled
+
+- ### Matthew Fredrickson (4):
+ - Revert "app_stack: Print proper exit location for PBXless channels."
+ - app_macro: Fix locking around datastore access
+ - app_followme.c: Grab reference on nativeformats before using it
+ - res_odbc.c: Allow concurrent access to request odbc connections
+
+- ### Matthew Kern (1):
+ - res_pjsip_t38: bind UDPTL sessions like RTP
+
+- ### Maximilian Fridrich (13):
+ - app_dial: Flip stream direction of outgoing channel.
+ - core_unreal: Flip stream direction of second channel.
+ - chan_pjsip: Only set default audio stream on hold.
+ - res_pjsip: Add 100rel option "peer_supported".
+ - res_pjsip: Add mediasec capabilities.
+ - core & res_pjsip: Improve topology change handling.
+ - res_pjsip: mediasec: Add Security-Client headers after 401
+ - chan_pjsip: Allow topology/session refreshes in early media state
+ - core/ari/pjsip: Add refer mechanism
+ - main/refer.c: Fix double free in refer_data_destructor + potential leak
+ - chan_rtp: Implement RTP glue for UnicastRTP channels
+ - app_dial: Add option "j" to preserve initial stream topology of caller
+ - res_pjsip_nat: Fix potential use of uninitialized transport details
+
+- ### Michael Cargile (1):
+ - apps/confbridge: Added hear_own_join_sound option to control who hears sound_j..
+
+- ### Michael Kuron (2):
+ - res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
+ - manager: AOC-S support for AOCMessage
+
+- ### Michael Neuhauser (2):
+ - pjproject: clone sdp to protect against (nat) modifications
+ - res_pjsip: delay contact pruning on Asterisk start
+
+- ### Michal Hajek (1):
+ - res_stasis.c: Add compare function for bridges moh container
+
+- ### Michał Górny (5):
+ - include: Remove unimplemented HMAC declarations
+ - BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD
+ - main: Enable rdtsc support on NetBSD
+ - main/utils: Implement ast_get_tid() for NetBSD
+ - build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD
+
+- ### Miguel Angel Nubla (1):
+ - configure: Makefile downloader enable follow redirects.
+
+- ### Mike Bradeen (44):
+ - build: prevent binary downloads for non x86 architectures
+ - various: Fix GCC 11 compilation issues.
+ - astobj2.c: Fix core when ref_log enabled
+ - res_rtp_asterisk: Addressing possible rtp range issues
+ - sched: fix and test a double deref on delete of an executing call back
+ - taskprocessor.c: Prevent crash on graceful shutdown
+ - Makefile: Avoid git-make user conflict
+ - CI: use Python3 virtual environment
+ - CI: Fixing path issue on venv check
+ - alembic: add missing ps_endpoints columns
+ - res_pjsip: Add user=phone on From and PAID for usereqphone=yes
+ - audiohook: add directional awareness
+ - res_pjsip: prevent crash on websocket disconnect
+ - ooh323c: not checking for IE minimum length
+ - manager: prevent file access outside of config dir
+ - app_directory: add ability to specify configuration file
+ - res_pjsip: Upgraded bundled pjsip to 2.13
+ - res_pjsip: Prevent SEGV in pjsip_evsub_send_request
+ - app_senddtmf: Add option to answer target channel.
+ - app_directory: Add a 'skip call' option.
+ - app_read: Add an option to return terminator on empty digits.
+ - res_pjsip_pubsub: subscription cleanup changes
+ - cli: increase channel column width
+ - format_sln: add .slin as supported file extension
+ - res_mixmonitor: MixMonitorMute by MixMonitor ID
+ - bridge_builtin_features: add beep via touch variable
+ - cel: add local optimization begin event
+ - indications: logging changes
+ - utils: add lock timestamps for DEBUG_THREADS
+ - res_musiconhold: avoid moh state access on unlocked chan
+ - app_voicemail: fix imap compilation errors
+ - app_voicemail: add CLI commands for message manipulation
+ - Adds manager actions to allow move/remove/forward individual messages in a par..
+ - app_voicemail: Fix for loop declarations
+ - res_pjsip: disable raw bad packet logging
+ - res_speech_aeap: check for null format on response
+ - func_periodic_hook: Add hangup step to avoid timeout
+ - cel: add publish user event helper
+ - res_speech_aeap: add aeap error handling
+ - res_pjsip: update qualify_timeout documentation with DNS note
+ - res_stasis: signal when new command is queued
+ - res_speech: allow speech to translate input channel
+ - app_voicemail: add NoOp alembic script to maintain sync
+ - app_chanspy: Add 'D' option for dual-channel audio
+
+- ### MikeNaso (1):
+ - res_pjsip.c: Set contact_user on incoming call local Contact header
+
+- ### Moritz Fain (1):
+ - ari: expose channel driver's unique id to ARI channel resource
+
+- ### Nathan Bruning (2):
+ - res_musiconhold: Don't crash when real-time doesn't return any entries
+ - app_queue: Add force_longest_waiting_caller option.
+
+- ### Naveen Albert (268):
+ - chan_sip: Expand hook flash recognition.
+ - main/file.c: Don't throw error on flash event.
+ - app_voicemail: Configurable voicemail beep
+ - AMI: Add AMI event to expose hook flash events
+ - func_volume: Add read capability to function.
+ - func_math: Three new dialplan functions
+ - sip_to_pjsip: Fix missing cases
+ - app_confbridge: New option to prevent answer supervision
+ - res_pjsip_dtmf_info: Hook flash
+ - app_confbridge: New ConfKick() application
+ - app_originate: Allow setting Caller ID and variables
+ - pbx_builtins: Corrects SayNumber warning
+ - app_dial: Expanded A option to add caller announcement
+ - app_waitforcond: New application
+ - app_reload: New Reload application
+ - app_dtmfstore: New application to store digits
+ - app_queue: Allow streaming multiple announcement files
+ - cdr_adaptive_odbc: Prevent filter warnings
+ - func_frame_drop: New function
+ - chan_alsa, chan_sip: Add replacement to moduleinfo
+ - app_originate: Add ability to set codecs
+ - func_scramble: Audio scrambler function
+ - app_morsecode: Add American Morse code
+ - func_math: Return integer instead of float if possible
+ - app_milliwatt: Timing fix
+ - app_queue: Don't reset queue stats on reload
+ - bridge_basic: Change warning to verbose if transfer cancelled
+ - app_read: Allow reading # as a digit
+ - chan_iax2: Add ANI2/OLI information element
+ - res_tonedetect: Tone detection module
+ - func_sayfiles: Retrieve say file names
+ - func_env: Add DIRNAME and BASENAME functions
+ - func_strings: Add STRBETWEEN function
+ - app_stack: Include current location if branch fails
+ - app_mf: Add channel agnostic MF sender
+ - res_pjsip_caller_id: Add ANI2/OLI parsing
+ - logger: Add custom logging capabilities
+ - app_queue: Fix hint updates for included contexts
+ - func_channel: Add CHANNEL_EXISTS function.
+ - func_vmcount: Add support for multiple mailboxes
+ - app_read: Fix null pointer crash
+ - chan_iax2: Add encryption for RSA authentication
+ - chan_iax2: Allow both secret and outkey at dial time
+ - app_voicemail: Fix phantom voicemail bug on rerecord
+ - sig_analog: Fix truncated buffer copy
+ - res_pjsip_callerid: Fix OLI parsing
+ - app_read: Fix custom terminator functionality regression
+ - app_morsecode: Fix deadlock
+ - res_tonedetect: Add call progress tone detection
+ - app_voicemail: Refactor email generation functions
+ - documentation: Standardize examples
+ - app_mf: Add full tech-agnostic MF support
+ - pbx: Add variable substitution API for extensions
+ - configs: Updates to sample configs
+ - func_json: Adds JSON_DECODE function
+ - chan_sip: Fix crash when accessing RURI before initiating outgoing call
+ - pbx_variables: Increase parsing capabilities of MSet
+ - strings: Fix enum names in comment examples
+ - app_sendtext: Add ReceiveText application
+ - app.c: Throw warnings for nonexistent options
+ - pbx_variables: initialize uninitialized variable
+ - app_sf: Add full tech-agnostic SF support
+ - documentation: Add missing AMI documentation
+ - app_mp3: Throw warning on nonexistent stream
+ - cli: Add module refresh command
+ - ami: Add AMI event for Wink
+ - dsp: Add define macro for DTMF_MATRIX_SIZE
+ - say.conf: fix 12pm noon logic
+ - pbx_variables: add missing ASTSBINDIR variable
+ - documentation: Document built-in system and channel vars
+ - res_rtp_asterisk: Fix typo in flag test/set
+ - frame.h: Fix spelling typo
+ - func_frame_drop: Fix typo referencing wrong buffer
+ - res_tonedetect: Fixes some logic issues and typos
+ - cdr: allow disabling CDR by default on new channels
+ - ami: Allow events to be globally disabled.
+ - app_mf: Add max digits option to ReceiveMF.
+ - app_mp3: Document and warn about HTTPS incompatibility.
+ - documentation: Adds missing default attributes.
+ - cli: Add core dump info to core show settings.
+ - func_db: Add validity check for key names when writing.
+ - app_voicemail: Emit warning if asking for nonexistent mailbox.
+ - res_stir_shaken: refactor utility function
+ - asterisk: Add macro for curl user agent.
+ - documentation: Add since tag to xmldocs DTD
+ - configs, LICENSE: remove pbx.digium.com.
+ - channel.c: Clean up debug level 1.
+ - func_channel: Add lastcontext and lastexten.
+ - ami: Improve substring parsing for disabled events.
+ - res_agi: Fix xmldocs bug with set music.
+ - pbx_builtins: Add missing options documentation
+ - app_dial: Document DIALSTATUS return values.
+ - chan_iax2: Fix perceived showing host address.
+ - chan_iax2: Fix spacing in netstats command
+ - pbx.c: Warn if there are too many includes in a context.
+ - build: Remove obsolete leftover build references.
+ - app_meetme: Emit warning if conference not found.
+ - app_queue: Add music on hold option to Queue.
+ - app_mf, app_sf: Return -1 if channel hangs up.
+ - samples: Remove obsolete sample configs.
+ - documentation: Adds versioning information.
+ - cli: Add command to evaluate dialplan functions.
+ - chan_pjsip: Add ability to send flash events.
+ - file.c: Prevent formats from seeking negative offsets.
+ - func_evalexten: Extension evaluation function.
+ - chan_dahdi: Fix insufficient array size for round robin.
+ - func_db: Add function to return cardinality at prefix
+ - asterisk.c: Warn of incompatibilities with remote console.
+ - app_meetme: Don't erroneously set global variables.
+ - menuselect: Don't erroneously recompile modules.
+ - chan_iax2: Prevent crash if dialing RSA-only call without outkey.
+ - chan_dahdi: Don't append cadences on dahdi restart.
+ - chan_dahdi: Don't allow MWI FSK if channel not idle.
+ - chan_dahdi: Document dial resource options.
+ - chan_dahdi: Fix broken operator mode clearing.
+ - app_confbridge: Add function to retrieve channels.
+ - res_pjsip_outbound_registration: Show time until expiration
+ - res_parking: Warn if out of bounds parking spot requested.
+ - res_calendar: Prevent assertion if event ends in past.
+ - loader: Prevent deadlock using tab completion.
+ - chan_iax2: Prevent deadlock due to duplicate autoservice.
+ - res_pjsip_outbound_registration: Make max random delay configurable.
+ - xmldocs: Improve examples.
+ - res_parking: Add music on hold override option.
+ - app_voicemail: Add option to prevent message deletion.
+ - sig_analog: Fix broken three-way conferencing.
+ - asterisk.c: Fix incompatibility warnings for remote console.
+ - pbx: Add helper function to execute applications.
+ - say: Abort play loop if caller hangs up.
+ - res_calendar_icalendar: Send user agent in request.
+ - app_dial: Propagate outbound hook flashes.
+ - cli: Fix CLI blocking forever on terminating backslash
+ - db: Notify user if deleted DB entry didn't exist.
+ - app_dial: Fix dial status regression.
+ - res_cliexec: Add dialplan exec CLI command.
+ - chan_iax2: Allow compiling without OpenSSL.
+ - general: Fix various typos.
+ - app_confbridge: Always set minimum video update interval.
+ - chan_dahdi: Add POLARITY function.
+ - chan_dahdi: Fix buggy and missing Caller ID parameters
+ - manager: Fix incomplete filtering of AMI events.
+ - db: Add AMI action to retrieve DB keys at prefix.
+ - pbx_functions.c: Manually update ast_str strlen.
+ - func_srv: Document field parameter.
+ - app_meetme: Add missing AMI documentation.
+ - app_confbridge: Add missing AMI documentation.
+ - general: Remove obsolete SVN references.
+ - cdr.conf: Remove obsolete app_mysql reference.
+ - general: Improve logging levels of some log messages.
+ - manager: Remove documentation for nonexistent action.
+ - app_confbridge: Fix memory leak on updated menu options.
+ - chan_iax2: Add missing options documentation.
+ - general: Very minor coding guideline fixes.
+ - features: Add transfer initiation options.
+ - res_tonedetect: Fix typos referring to wrong variables.
+ - cli: Prevent assertions on startup from bad ao2 refs.
+ - pbx_variables: Use const char if possible.
+ - app_confbridge: Add end_marked_any option.
+ - lock.c: Add AMI event for deadlocks.
+ - func_frame_trace: Remove bogus assertion.
+ - func_strings: Add trim functions.
+ - func_scramble: Fix null pointer dereference.
+ - func_export: Add EXPORT function
+ - app_amd: Add option to play audio during AMD.
+ - app_bridgewait: Add option to not answer channel.
+ - features: Add no answer option to Bridge.
+ - func_logic: Don't emit warning if both IF branches are empty.
+ - db: Fix incorrect DB tree count for AMI.
+ - res_pjsip_geolocation: Change some notices to debugs.
+ - chan_dahdi: Resolve format truncation warning.
+ - res_tonedetect: Add ringback support to TONE_DETECT.
+ - cdr: Allow bridging and dial state changes to be ignored.
+ - say: Don't prepend ampersand erroneously.
+ - res_pjsip_pubsub: Prevent removing subscriptions.
+ - chan_dahdi: Fix unavailable channels returning busy.
+ - res_pjsip_logger: Add method-based logging option.
+ - res_pjsip_notify: Add option support for AMI.
+ - tests: Fix compilation errors on 32-bit.
+ - tcptls: Prevent crash when freeing OpenSSL errors.
+ - app_stack: Print proper exit location for PBXless channels.
+ - file.c: Don't emit warnings on winks.
+ - manager: Update ModuleCheck documentation.
+ - app_mixmonitor: Add option to delete files on exit.
+ - test_json: Remove duplicated static function.
+ - func_json: Fix memory leak.
+ - sla: Prevent deadlock and crash due to autoservicing.
+ - chan_dahdi: Allow FXO channels to start immediately.
+ - pbx_builtins: Allow Answer to return immediately.
+ - app_mixmonitor: Add option to use real Caller ID for voicemail.
+ - rtp_engine.h: Update examples using ast_format_set.
+ - xmldoc: Allow XML docs to be reloaded.
+ - sig_analog: Fix no timeout duration.
+ - func_presencestate: Fix invalid memory access.
+ - res_pjsip_header_funcs: Add custom parameter support.
+ - res_adsi: Fix major regression caused by media format rearchitecture.
+ - res_pjsip_session.c: Map empty extensions in INVITEs to s.
+ - app_if: Adds conditional branch applications
+ - res_hep: Add support for named capture agents.
+ - app_voicemail: Fix missing email in msg_create_from_file.
+ - app_if: Fix format truncation errors.
+ - app_sendtext: Remove references to removed applications.
+ - func_callerid: Warn about invalid redirecting reason.
+ - app_voicemail_odbc: Fix string overflow warning.
+ - res_pjsip_session: Use Caller ID for extension matching.
+ - pbx_app: Update outdated pbx_exec channel snapshots.
+ - manager: Fix appending variables.
+ - json.h: Add ast_json_object_real_get.
+ - func_frame_trace: Print text for text frames.
+ - app_broadcast: Add Broadcast application
+ - loader: Allow declined modules to be unloaded.
+ - res_pjsip_session: Add overlap_context option.
+ - func_json: Enhance parsing capabilities of JSON_DECODE
+ - app_signal: Add signaling applications
+ - chan_iax2: Fix jitterbuffer regression prior to receiving audio.
+ - app_dial: Fix DTMF not relayed to caller on unanswered calls.
+ - app_senddtmf: Add SendFlash AMI action.
+ - func_json: Fix JSON parsing issues.
+ - app_queue: Fix minor xmldoc duplication and vagueness.
+ - voicemail.conf: Fix incorrect comment about #include.
+ - pbx_dundi: Add PJSIP support.
+ - res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
+ - chan_dahdi: Add dialmode option for FXS lines.
+ - asterisk.c: Fix option warning for remote console.
+ - chan_dahdi: Fix broken hidecallerid setting.
+ - logrotate: Fix duplicate log entries.
+ - callerid: Allow specifying timezone for date/time.
+ - sig_analog: Add fuller Caller ID support.
+ - chan_dahdi: Fix Caller ID presentation for FXO ports.
+ - res_musiconhold: Add option to loop last file.
+ - sig_analog: Allow immediate fake ring to be suppressed.
+ - sig_analog: Allow three-way flash to time out to silence.
+ - chan_dahdi: Allow autoreoriginating after hangup.
+ - res_pjsip_header_funcs: Make prefix argument optional.
+ - sig_analog: Add Called Subscriber Held capability.
+ - pbx.c: Fix gcc 12 compiler warning.
+ - app_dial: Fix infinite loop when sending digits.
+ - chan_iax2: Improve authentication debugging.
+ - chan_console: Fix deadlock caused by unclean thread exit.
+ - app_voicemail: Disable ADSI if unavailable.
+ - chan_dahdi: Clarify scope of callgroup/pickupgroup.
+ - res_pjsip: Include cipher limit in config error message.
+ - app_voicemail: Add AMI event for mailbox PIN changes.
+ - core_local: Fix local channel parsing with slashes.
+ - app_directory: Add ADSI support to Directory.
+ - chan_dahdi: Warn if nonexistent cadence is requested.
+ - logger: Add channel-based filtering.
+ - configs: Improve documentation for bandwidth in iax.conf.
+ - func_lock: Add missing see-also refs to documentation.
+ - func_channel: Expose previously unsettable options.
+ - sig_analog: Fix channel leak when mwimonitor is enabled.
+ - general: Fix broken links.
+ - manager.c: Improve clarity of "manager show connected".
+ - config_options.c: Fix truncation of option descriptions.
+ - manager.c: Fix regression due to using wrong free function.
+ - app_if: Fix faulty EndIf branching.
+ - menuselect: Use more specific error message.
+ - logger: Fix linking regression.
+ - func_frame_trace: Add CLI command to dump frame queue.
+ - chan_dahdi: Allow MWI to be manually toggled on channels.
+ - res_calendar_icalendar: Print iCalendar error on parsing failure.
+ - manager.c: Fix erroneous reloads in UpdateConfig.
+ - app_if: Fix next priority calculation.
+ - configure: Rerun bootstrap on modern platform.
+ - dsp.c: Fix and improve potentially inaccurate log message.
+ - app_voicemail: Allow preventing mark messages as urgent.
+ - app_voicemail: Properly reinitialize config after unit tests.
+ - app_dial: Add dial time for progress/ringing.
+ - pbx_variables.c: Prevent SEGV due to stack overflow.
+
+- ### Nick French (4):
+ - res_pjsip_session: Restore calls to ast_sip_message_apply_transport()
+ - res_pjsip: Prevent segfault in UDP registration with flow transports
+ - res_pjsip: dont return early from registration if init auth fails
+ - pjproject_bundled: Fix cross-compilation with SSL libs.
+
+- ### Nickolay Shmyrev (1):
+ - res_speech: Bump reference on format object
+
+- ### Nico Kooijman (1):
+ - main: With Dutch language year after 2020 is not spoken in say.c
+
+- ### Niklas Larsson (1):
+ - app_queue: Preserve reason for realtime queues
+
+- ### Olaf Titz (1):
+ - app_voicemail_imap: Fix message count when IMAP server is unavailable
+
+- ### Patrick Verzele (1):
+ - res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a..
+
+- ### Peter Fern (1):
+ - streams: Ensure that stream is closed in ast_stream_and_wait on error
+
+- ### PeterHolik (2):
+ - chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
+ - chan_rtp.c: MulticastRTP missing refcount without codec option
+
+- ### Philip Prindeville (14):
+ - logger: workaround woefully small BUFSIZ in MUSL
+ - time: add support for time64 libcs
+ - res_crypto: Don't load non-regular files in keys directory
+ - test: Add ability to capture child process output
+ - main/utils: allow checking for command in $PATH
+ - test: Add test coverage for capture child process output
+ - res_crypto: make keys reloadable on demand for testing
+ - test: Add coverage for res_crypto
+ - res_crypto: Use EVP API's instead of legacy API's
+ - res_crypto: don't complain about directories
+ - test: initialize capture structure before freeing
+ - res_crypto: use ast_file_read_dirs() to iterate
+ - res_crypto: don't modify fname in try_load_key()
+ - res_crypto: handle unsafe private key files
+
+- ### Pirmin Walthert (1):
+ - res_pjsip_nat.c: Create deep copies of strings when appropriate
+
+- ### Richard Mudgett (2):
+ - res_pjsip_session.c: Fix compiler warnings.
+ - chan_vpb.cc: Fix compile errors.
+
+- ### Rijnhard Hessel (1):
+ - res_statsd: handle non-standard meter type safely
+
+- ### Robert Cripps (1):
+ - res/res_pjsip_session.c: Check that media type matches in function ast_sip_ses..
+
+- ### Rodrigo Ramírez Norambuena (1):
+ - app_queue: Add LoginTime field for member in a queue.
+
+- ### Salah Ahmed (1):
+ - res_rtp_asterisk: Check remote ICE reset and reset local ice attrb
+
+- ### Sam Banks (1):
+ - queues.conf.sample: Correction of typo
+
+- ### Samuel Olaechea (1):
+ - configs: Fix typo in pjsip.conf.sample.
+
+- ### Sarah Autumn (1):
+ - sig_analog: Changes to improve electromechanical signalling compatibility
+
+- ### Sean Bright (155):
+ - acl.c: Coerce a NULL pointer into the empty string
+ - vector.h: Add AST_VECTOR_SORT()
+ - utf8.c: Add UTF-8 validation and utility functions
+ - vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
+ - res_musiconhold.c: Prevent crash with realtime MoH
+ - res_musiconhold.c: Use ast_file_read_dir to scan MoH directory
+ - bridge_channel: Ensure text messages are zero terminated
+ - app_voicemail: Process urgent messages with mailcmd
+ - format_cap: Perform codec lookups by pointer instead of name
+ - res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined
+ - audiosocket: Fix module menuselect descriptions
+ - chan_sip.c: Don't build by default
+ - res_musiconhold: Start playlist after initial announcement
+ - func_curl.c: Prevent crash when using CURLOPT(httpheader)
+ - dsp.c: Update calls to ast_format_cmp to check result properly
+ - res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs
+ - app_voicemail.c: Document VMSayName interruption behavior
+ - pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
+ - tcptls.c: Don't close TCP client file descriptors more than once
+ - features.conf.sample: Sample sound files incorrectly quoted
+ - sip_to_pjsip.py: Handle #include globs and other fixes
+ - CHANGES: Remove already applied CHANGES update
+ - media_cache: Fix reference leak with bucket file metadata
+ - res_http_media_cache.c: Set reasonable number of redirects
+ - app_chanspy: Spyee information missing in ChanSpyStop AMI Event
+ - asterisk: Export additional manager functions
+ - app_voicemail: Prevent deadlocks when out of ODBC database connections
+ - res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet
+ - app_read: Release tone zone reference on early return.
+ - res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
+ - app_page.c: Don't fail to Page if beep sound file is missing
+ - res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
+ - strings.h: ast_str_to_upper() and _to_lower() are not pure.
+ - modules.conf: Fix differing usage of assignment operators.
+ - app_dial.c: Only send DTMF on first progress event.
+ - app_queue.c: Don't crash when realtime queue name is empty.
+ - queues.conf.sample: Correct 'context' documentation.
+ - app_queue.c: Remove dead 'updatecdr' code.
+ - modules.conf: Fix more differing usages of assignment operators.
+ - app_queue: Add alembic migration to add ringinuse to queue_members.
+ - loader.c: Speed up deprecation metadata lookup
+ - res_pjsip.c: OPTIONS processing can now optionally skip authentication
+ - res_rtp_asterisk: More robust timestamp checking
+ - translate.c: Avoid refleak when checking for a translation path
+ - chan_pjsip: Correct misleading trace message
+ - menuselect: Fix description of several modules.
+ - res_pjsip_config_wizard.c: Add port matching support.
+ - main/cdr.c: Correct Party A selection.
+ - res_http_media_cache.c: Parse media URLs to find extensions.
+ - res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.
+ - res_http_media_cache.c: Fix merge errors from 18 -> master
+ - res_http_media_cache: Cleanup audio format lookup in HTTP requests
+ - mgcp: Remove dead debug code
+ - config_options: Handle ACO arrays correctly in generated XML docs.
+ - dns.c: Load IPv6 DNS resolvers if configured.
+ - term.c: Add support for extended number format terminfo files.
+ - app_voicemail.c: Ability to silence instructions if greeting is present.
+ - test_abstract_jb.c: Fix put and put_out_of_order memory leaks.
+ - test_http_media_cache.c: Fix copy/paste error during test deregistration.
+ - app_externalivr.c: Fix mixed leading whitespace in source code.
+ - res_http_media_cache.c: Compare unaltered MIME types.
+ - message.c: Support 'To' header override with AMI's MessageSend.
+ - Makefile: Use basename in a POSIX-compliant way.
+ - configure: Remove unused OpenSSL SRTP check.
+ - func_talkdetect.c: Fix logical errors in silence detection.
+ - various: Fix GCC 11.2 compilation issues.
+ - pbx.c: Don't remove dashes from hints on reload.
+ - config.c: Prevent UB in ast_realtime_require_field.
+ - channel: Short-circuit ast_channel_get_by_name() on empty arg.
+ - CHANGES: Correct reference to configuration file.
+ - say.c: Honor requests for DTMF interruption.
+ - pjproject: Fix incorrect unescaping of tokens during parsing
+ - utils.c: Remove all usages of ast_gethostbyname()
+ - say.c: Prevent erroneous failures with 'say' family of functions.
+ - build: Rebuild configure and autoconfig.h.in
+ - build_tools/make_version: Fix bashism in comparison.
+ - manager.c: Generate valid XML if attribute names have leading digits.
+ - res_pjsip.c: Correct minor typos in 'realm' documentation.
+ - manager.c: Simplify AMI ModuleCheck handling
+ - conversions.c: Specify that we only want to parse decimal numbers.
+ - download_externals: Use HTTPS for downloads
+ - stasis_recording: Perform a complete match on requested filename.
+ - openssl: Supress deprecation warnings from OpenSSL 3.0
+ - config.h: Don't use C++ keywords as argument names.
+ - loader.c: Use portable printf conversion specifier for int64.
+ - ast_pkgconfig.m4: AST_PKG_CONFIG_CHECK() relies on sed.
+ - pbx.c: Simplify ast_context memory management.
+ - channel.h: Remove redundant declaration.
+ - chan_dahdi.c: Resolve a format-truncation build warning.
+ - app_playback.c: Fix PLAYBACKSTATUS regression.
+ - pbx_ael: Global variables are not expanded.
+ - doxygen: Fix doxygen errors.
+ - app_queue: Reset all queue defaults before reload.
+ - app_queue: Minor docs and logging fixes for UnpauseQueueMember.
+ - test_crypto.c: Fix getcwd(…) build error.
+ - test.c: Avoid passing -1 to FD_* family of functions.
+ - Revert "pbx_ael: Global variables are not expanded."
+ - contrib: rc.archlinux.asterisk uses invalid redirect.
+ - res_agi: RECORD FILE plays 2 beeps.
+ - ael: Regenerate lexers and parsers.
+ - loader.c: Minor module key check simplification.
+ - core: Cleanup gerrit and JIRA references. (#57)
+ - apply_patches: Use globbing instead of file/sort.
+ - xml.c: Process XML Inclusions recursively.
+ - res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
+ - sounds: Update download URL to use HTTPS.
+ - ast-db-manage: Fix alembic branching error caused by #122.
+ - res_crypto.c: Avoid using the non-portable ALLPERMS macro.
+ - ast-db-manage: Synchronize revisions between comments and code.
+ - configure: Remove obsolete and deprecated constructs.
+ - res_crypto.c: Gracefully handle potential key filename truncation.
+ - pjsip_transport_events.c: Use %zu printf specifier for size_t.
+ - res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
+ - res_geolocation: Ensure required 'location_info' is present.
+ - chan_iax2.c: Avoid crash with IAX2 switch support.
+ - func_export: Use correct function argument as variable name.
+ - extensions.conf.sample: Remove reference to missing context.
+ - res_pjsip: Enable TLS v1.3 if present.
+ - extconfig: Allow explicit DB result set ordering to be disabled.
+ - res_stasis_recording.c: Save recording state when unmuted.
+ - func_curl.c: Ensure channel is locked when manipulating datastores.
+ - pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
+ - res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
+ - res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
+ - app_queue.c: Emit unpause reason with PauseQueueMember event.
+ - chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
+ - doc: Update IP Quality of Service links.
+ - resource_channels.c: Explicit codec request when creating UnicastRTP.
+ - chan_iax2.c: Don't send unsanitized data to the logger.
+ - live_ast: Add astcachedir to generated asterisk.conf.
+ - res_http_websocket.c: Set hostname on client for certificate validation.
+ - func_curl.c: Remove CURLOPT() plaintext documentation.
+ - uri.c: Simplify ast_uri_make_host_with_port()
+ - app.c: Allow ampersands in playback lists to be escaped.
+ - alembic: Update list of TLS methods available on ps_transports.
+ - res_rtp_asterisk.c: Update for OpenSSL 3+.
+ - app_voicemail.c: Completely resequence mailbox folders.
+ - make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
+ - config.c: Log #exec include failures.
+ - res_pjsip_header_funcs.c: Check URI parameter length before copying.
+ - logger.c: Move LOG_GROUP documentation to dedicated XML file.
+ - make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
+ - app_confbridge: Don't emit warnings on valid configurations.
+ - rtp_engine.c: Correct sample rate typo for L16/44100.
+ - res_pjsip_session.c: Correctly format SDP connection addresses.
+ - res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
+ - strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
+ - alembic: Synchronize alembic heads between supported branches.
+ - res_pjsip: Use consistent type for boolean columns.
+ - alembic: Correct NULLability of PJSIP id columns.
+ - alembic: Quote new MySQL keyword 'qualify.'
+ - res_pjsip: Fix alembic downgrade for boolean columns.
+ - alembic: Fix compatibility with SQLAlchemy 2.0+.
+ - xml.c: Update deprecated libxml2 API usage.
+ - logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
+
+- ### Sebastian Jennen (1):
+ - translate.c: implement new direct comp table mode
+
+- ### Sebastien Duthil (4):
+ - app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute.
+ - stun: Emit warning message when STUN request times out
+ - res_rtp_asterisk: Automatically refresh stunaddr from DNS
+ - main/stun.c: fix crash upon STUN request timeout
+
+- ### Sergey V. Lobanov (1):
+ - build: fix bininstall launchd issue on cross-platform build
+
+- ### Shaaah (1):
+ - app_queue.c : fix "queue add member" usage string
+
+- ### Shloime Rosenblum (3):
+ - main/say.c: Support future dates with Q and q format params
+ - apps/app_playback.c: Add 'mix' option to app_playback
+ - res_agi: Evaluate dialplan functions and variables in agi exec if enabled
+
+- ### Shyju Kanaprath (1):
+ - README.md: Removed outdated link
+
+- ### Spiridonov Dmitry (1):
+ - sorcery.c: Fixed crash error when executing "module reload"
+
+- ### Stanislav (1):
+ - res_pjsip_stir_shaken: Fix module description
+
+- ### Stanislav Abramenkov (2):
+ - pjsip: Upgrade bundled version to pjproject 2.12.1
+ - pjsip: Upgrade bundled version to pjproject 2.13.1
+
+- ### Steve Davies (1):
+ - app_queue: Fix hint updates, allow dup. hints
+
+- ### Sungtae Kim (5):
+ - res_stasis.c: Added video_single option for bridge creation
+ - realtime: Increased reg_server character size
+ - res_ari: Fix wrong media uri handle for channel play
+ - res_pjsip_session: Fixed NULL active media topology handle
+ - resource_channels.c: Fix external media data option
+
+- ### The_Blode (1):
+ - install_prereq: Add Linux Mint support.
+
+- ### Thomas Guebels (1):
+ - res_pjsip_transport_websocket: save the original contact host
+
+- ### Tinet-mucw (1):
+ - res_pjsip_transport_websocket: Prevent transport from being destroyed before m..
+
+- ### Torrey Searle (6):
+ - res_pjsip_diversion: handle 181
+ - res_pjsip_diversion: implement support for History-Info
+ - res_pjsip_diversion: fix double 181
+ - res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
+ - res/res_rtp_asterisk: generate new SSRC on native bridge end
+ - res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf
+
+- ### Trevor Peirce (2):
+ - res_pjsip: Actually enable session timers when timers=always
+ - features: Update documentation for automon and automixmon
+
+- ### Vitezslav Novy (1):
+ - res_rtp_asterisk: fix wrong counter management in ioqueue objects
+
+- ### Walter Doekes (1):
+ - main/say: Work around gcc 9 format-truncation false positive
+
+- ### Yury Kirsanov (1):
+ - bridge_simple.c: Unhold channels on join simple bridge.
+
+- ### alex2grad (1):
+ - app_followme: fix issue with enable_callee_prompt=no (#88)
+
+- ### cmaj (3):
+ - Makefile: Fix certified version numbers
+ - res_phoneprov.c: Multihomed SERVER cache prevention
+ - app_speech_utils.c: Allow partial speech results.
+
+- ### lvl (2):
+ - res_musiconhold: Load all realtime entries, not just the first
+ - Introduce astcachedir, to be used for temporary bucket files
+
+- ### phoneben (1):
+ - func_cut: Add example to documentation.
+
+- ### roadkill (1):
+ - res/res_pjsip.c: allow user=phone when number contain *#
+
+- ### romryz (1):
+ - res_rtp_asterisk.c: Correct coefficient in MOS calculation.
+
+- ### sungtae kim (5):
+ - stasis_bridge.c: Fixed wrong video_mode shown
+ - resource_channels.c: Fix wrong external media parameter parse
+ - res_musiconhold: Add option to not play music on hold on unanswered channels
+ - res_stasis_snoop: Fix snoop crash
+ - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..
+
+- ### under (1):
+ - codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother
+
+- ### zhengsh (3):
+ - res_sorcery_memory_cache.c: Fix memory leak
+ - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` i..
+ - app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+
+- ### zhou_jiajian (1):
+ - res_fax_spandsp.c: Clean up a spaces/tabs issue
+
+
+### Commit List:
+
+- logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
+- app_voicemail_odbc: Allow audio to be kept on disk
+- security_agreement.c: Always add the Require and Proxy-Require headers
+- stasis_channels: Use uniqueid and name to delete old snapshots
+- xml.c: Update deprecated libxml2 API usage.
+- stir_shaken: Fix memory leak, typo in config, tn canonicalization
+- sorcery.c: Fixed crash error when executing "module reload"
+- logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
+- res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2)
+- res_ari.c: Add additional output to ARI requests when debug is enabled
+- alembic: Fix compatibility with SQLAlchemy 2.0+.
+- pbx_variables.c: Prevent SEGV due to stack overflow.
+- res_pjsip: Fix alembic downgrade for boolean columns.
+- alembic: Quote new MySQL keyword 'qualify.'
+- asterisk.c: Fix sending incorrect messages to systemd notify
+- tcptls/iostream: Add support for setting SNI on client TLS connections
+- make_buildopts_h: Always include DETECT_DEADLOCKS
+- alembic: Correct NULLability of PJSIP id columns.
+- rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
+- manager.c: Add missing parameters to Login documentation
+- Fix incorrect application and function documentation references
+- Initial commit for certified-20.7
+- res_pjsip_stir_shaken.c: Add checks for missing parameters
+- app_dial: Add dial time for progress/ringing.
+- app_voicemail: Properly reinitialize config after unit tests.
+- app_queue.c : fix "queue add member" usage string
+- app_voicemail: Allow preventing mark messages as urgent.
+- res_pjsip: Use consistent type for boolean columns.
+- attestation_config.c: Use ast_free instead of ast_std_free
+- Makefile: Add stir_shaken/cache to directories created on install
+- Stir/Shaken Refactor
+- alembic: Synchronize alembic heads between supported branches.
+- translate.c: implement new direct comp table mode
+- README.md: Removed outdated link
+- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
+- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
+- dsp.c: Fix and improve potentially inaccurate log message.
+- pjsip show channelstats: Prevent possible segfault when faxing
+- Reduce startup/shutdown verbose logging
+- configure: Rerun bootstrap on modern platform.
+- Upgrade bundled pjproject to 2.14.
+- app_speech_utils.c: Allow partial speech results.
+- utils: Make behavior of ast_strsep* match strsep.
+- app_chanspy: Add 'D' option for dual-channel audio
+- app_if: Fix next priority calculation.
+- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
+- BuildSystem: Bump autotools versions on OpenBSD.
+- main/utils: Simplify the FreeBSD ast_get_tid() handling
+- res_pjsip_session.c: Correctly format SDP connection addresses.
+- rtp_engine.c: Correct sample rate typo for L16/44100.
+- manager.c: Fix erroneous reloads in UpdateConfig.
+- res_calendar_icalendar: Print iCalendar error on parsing failure.
+- app_confbridge: Don't emit warnings on valid configurations.
+- app_voicemail: add NoOp alembic script to maintain sync
+- chan_dahdi: Allow MWI to be manually toggled on channels.
+- chan_rtp.c: MulticastRTP missing refcount without codec option
+- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
+- func_frame_trace: Add CLI command to dump frame queue.
+- logger: Fix linking regression.
+- Revert "core & res_pjsip: Improve topology change handling."
+- menuselect: Use more specific error message.
+- res_pjsip_nat: Fix potential use of uninitialized transport details
+- app_if: Fix faulty EndIf branching.
+- manager.c: Fix regression due to using wrong free function.
+- config_options.c: Fix truncation of option descriptions.
+- manager.c: Improve clarity of "manager show connected".
+- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
+- general: Fix broken links.
+- MergeApproved.yml: Remove unneeded concurrency
+- app_dial: Add option "j" to preserve initial stream topology of caller
+- ast_coredumper: Increase reliability
+- logger.c: Move LOG_GROUP documentation to dedicated XML file.
+- res_odbc.c: Allow concurrent access to request odbc connections
+- res_pjsip_header_funcs.c: Check URI parameter length before copying.
+- config.c: Log #exec include failures.
+- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
+- app_voicemail.c: Completely resequence mailbox folders.
+- sig_analog: Fix channel leak when mwimonitor is enabled.
+- res_rtp_asterisk.c: Update for OpenSSL 3+.
+- alembic: Update list of TLS methods available on ps_transports.
+- func_channel: Expose previously unsettable options.
+- app.c: Allow ampersands in playback lists to be escaped.
+- uri.c: Simplify ast_uri_make_host_with_port()
+- func_curl.c: Remove CURLOPT() plaintext documentation.
+- res_http_websocket.c: Set hostname on client for certificate validation.
+- live_ast: Add astcachedir to generated asterisk.conf.
+- SECURITY.md: Update with correct documentation URL
+- func_lock: Add missing see-also refs to documentation.
+- app_followme.c: Grab reference on nativeformats before using it
+- configs: Improve documentation for bandwidth in iax.conf.
+- logger: Add channel-based filtering.
+- chan_iax2.c: Don't send unsanitized data to the logger.
+- codec_ilbc: Disable system ilbc if version >= 3.0.0
+- resource_channels.c: Explicit codec request when creating UnicastRTP.
+- doc: Update IP Quality of Service links.
+- chan_pjsip: Add PJSIPHangup dialplan app and manager action
+- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
+- chan_dahdi: Warn if nonexistent cadence is requested.
+- stasis: Update the snapshot after setting the redirect
+- ari: Provide the caller ID RDNIS for the channels
+- main/utils: Implement ast_get_tid() for OpenBSD
+- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
+- app_directory: Add ADSI support to Directory.
+- core_local: Fix local channel parsing with slashes.
+- Remove files that are no longer updated
+- app_voicemail: Add AMI event for mailbox PIN changes.
+- app_queue.c: Emit unpause reason with PauseQueueMember event.
+- bridge_simple: Suppress unchanged topology change requests
+- res_pjsip: Include cipher limit in config error message.
+- res_speech: allow speech to translate input channel
+- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
+- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
+- api.wiki.mustache: Fix indentation in generated markdown
+- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
+- configs: Fix typo in pjsip.conf.sample.
+- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
+- res_stasis: signal when new command is queued
+- ari/stasis: Indicate progress before playback on a bridge
+- func_curl.c: Ensure channel is locked when manipulating datastores.
+- Update config.yml
+- logger.h: Add ability to change the prefix on SCOPE_TRACE output
+- Add libjwt to third-party
+- res_pjsip: update qualify_timeout documentation with DNS note
+- chan_dahdi: Clarify scope of callgroup/pickupgroup.
+- func_json: Fix crashes for some types
+- res_speech_aeap: add aeap error handling
+- app_voicemail: Disable ADSI if unavailable.
+- codec_builtin: Use multiples of 20 for maximum_ms
+- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
+- asterisk.c: Use the euid's home directory to read/write cli history
+- cel: add publish user event helper
+- chan_console: Fix deadlock caused by unclean thread exit.
+- file.c: Add ability to search custom dir for sounds
+- chan_iax2: Improve authentication debugging.
+- res_rtp_asterisk: fix wrong counter management in ioqueue objects
+- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
+- func_periodic_hook: Add hangup step to avoid timeout
+- res_stasis_recording.c: Save recording state when unmuted.
+- res_speech_aeap: check for null format on response
+- func_periodic_hook: Don't truncate channel name
+- safe_asterisk: Change directory permissions to 755
+- chan_rtp: Implement RTP glue for UnicastRTP channels
+- app_queue: periodic announcement configurable start time.
+- variables: Add additional variable dialplan functions.
+- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
+- res_rtp_asterisk: Fix regression issues with DTLS client check
+- res_pjsip_header_funcs: Duplicate new header value, don't copy.
+- res_pjsip: disable raw bad packet logging
+- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+- manager.c: Prevent path traversal with GetConfig.
+- ari-stubs: Fix more local anchor references
+- ari-stubs: Fix more local anchor references
+- ari-stubs: Fix broken documentation anchors
+- res_pjsip_session: Send Session Interval too small response
+- app_dial: Fix infinite loop when sending digits.
+- app_voicemail: Fix for loop declarations
+- alembic: Fix quoting of the 100rel column
+- pbx.c: Fix gcc 12 compiler warning.
+- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+- download_externals: Fix a few version related issues
+- main/refer.c: Fix double free in refer_data_destructor + potential leak
+- sig_analog: Add Called Subscriber Held capability.
+- app_macro: Fix locking around datastore access
+- Revert "app_stack: Print proper exit location for PBXless channels."
+- install_prereq: Fix dependency install on aarch64.
+- res_pjsip.c: Set contact_user on incoming call local Contact header
+- extconfig: Allow explicit DB result set ordering to be disabled.
+- rest-api: Run make ari-stubs
+- res_pjsip_header_funcs: Make prefix argument optional.
+- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+- manager: Tolerate stasis messages with no channel snapshot.
+- core/ari/pjsip: Add refer mechanism
+- chan_dahdi: Allow autoreoriginating after hangup.
+- audiohook: Unlock channel in mute if no audiohooks present.
+- sig_analog: Allow three-way flash to time out to silence.
+- res_prometheus: Do not generate broken metrics
+- res_pjsip: Enable TLS v1.3 if present.
+- func_cut: Add example to documentation.
+- extensions.conf.sample: Remove reference to missing context.
+- func_export: Use correct function argument as variable name.
+- app_queue: Add support for applying caller priority change immediately.
+- chan_iax2.c: Avoid crash with IAX2 switch support.
+- res_geolocation: Ensure required 'location_info' is present.
+- app_voicemail: add CLI commands for message manipulation
+- sig_analog: Allow immediate fake ring to be suppressed.
+- app.h: Move declaration of ast_getdata_result before its first use
+- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
+- app_voicemail: fix imap compilation errors
+- res_musiconhold: avoid moh state access on unlocked chan
+- utils: add lock timestamps for DEBUG_THREADS
+- rest-api: Updates for new documentation site
+- app_voicemail_imap: Fix message count when IMAP server is unavailable
+- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
+- res_pjsip_session: Added new function calls to avoid ABI issues.
+- app_queue: Add force_longest_waiting_caller option.
+- pjsip_transport_events.c: Use %zu printf specifier for size_t.
+- res_crypto.c: Gracefully handle potential key filename truncation.
+- configure: Remove obsolete and deprecated constructs.
+- res_fax_spandsp.c: Clean up a spaces/tabs issue
+- ast-db-manage: Synchronize revisions between comments and code.
+- test_statis_endpoints: Fix channel_messages test again
+- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
+- tcptls: when disabling a server port, we should set the accept_fd to -1.
+- AMI: Add parking position parameter to Park action
+- test_stasis_endpoints.c: Make channel_messages more stable
+- build: Fix a few gcc 13 issues
+- ast-db-manage: Fix alembic branching error caused by #122.
+- app_followme: fix issue with enable_callee_prompt=no (#88)
+- sounds: Update download URL to use HTTPS.
+- configure: Makefile downloader enable follow redirects.
+- res_musiconhold: Add option to loop last file.
+- chan_dahdi: Fix Caller ID presentation for FXO ports.
+- AMI: Add CoreShowChannelMap action.
+- sig_analog: Add fuller Caller ID support.
+- res_stasis.c: Add new type 'sdp_label' for bridge creation.
+- app_queue: Preserve reason for realtime queues
+- indications: logging changes
+- callerid: Allow specifying timezone for date/time.
+- logrotate: Fix duplicate log entries.
+- chan_pjsip: Allow topology/session refreshes in early media state
+- chan_dahdi: Fix broken hidecallerid setting.
+- asterisk.c: Fix option warning for remote console.
+- configure: fix test code to match gethostbyname_r prototype.
+- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
+- res_sorcery_memory_cache.c: Fix memory leak
+- xml.c: Process XML Inclusions recursively.
+- apply_patches: Use globbing instead of file/sort.
+- apply_patches: Sort patch list before applying
+- pjsip: Upgrade bundled version to pjproject 2.13.1
+- Set up new ChangeLogs directory
+- chan_pjsip: also return all codecs on empty re-INVITE for late offers
+- cel: add local optimization begin event
+- core: Cleanup gerrit and JIRA references. (#57)
+- res_pjsip: mediasec: Add Security-Client headers after 401
+- LICENSE: Update link to trademark policy.
+- chan_dahdi: Add dialmode option for FXS lines.
+- Initial GitHub PRs
+- Initial GitHub Issue Templates
+- pbx_dundi: Fix PJSIP endpoint configuration check.
+- Revert "app_queue: periodic announcement configurable start time."
+- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
+- pbx_dundi: Add PJSIP support.
+- install_prereq: Add Linux Mint support.
+- chan_pjsip: fix music on hold continues after INVITE with replaces
+- voicemail.conf: Fix incorrect comment about #include.
+- app_queue: Fix minor xmldoc duplication and vagueness.
+- test.c: Fix counting of tests and add 2 new tests
+- res_calendar: output busy state as part of show calendar.
+- loader.c: Minor module key check simplification.
+- ael: Regenerate lexers and parsers.
+- bridge_builtin_features: add beep via touch variable
+- res_mixmonitor: MixMonitorMute by MixMonitor ID
+- format_sln: add .slin as supported file extension
+- res_agi: RECORD FILE plays 2 beeps.
+- func_json: Fix JSON parsing issues.
+- app_senddtmf: Add SendFlash AMI action.
+- app_dial: Fix DTMF not relayed to caller on unanswered calls.
+- configure: fix detection of re-entrant resolver functions
+- cli: increase channel column width
+- app_queue: periodic announcement configurable start time.
+- make_version: Strip svn stuff and suppress ref HEAD errors
+- res_http_media_cache: Introduce options and customize
+- main/iostream.c: fix build with libressl
+- contrib: rc.archlinux.asterisk uses invalid redirect.
+- res_pjsip_pubsub: subscription cleanup changes
+- Revert "pbx_ael: Global variables are not expanded."
+- res_pjsip: Replace invalid UTF-8 sequences in callerid name
+- test.c: Avoid passing -1 to FD_* family of functions.
+- chan_iax2: Fix jitterbuffer regression prior to receiving audio.
+- test_crypto.c: Fix getcwd(…) build error.
+- pjproject_bundled: Fix cross-compilation with SSL libs.
+- app_read: Add an option to return terminator on empty digits.
+- res_phoneprov.c: Multihomed SERVER cache prevention
+- app_directory: Add a 'skip call' option.
+- app_senddtmf: Add option to answer target channel.
+- res_pjsip: Prevent SEGV in pjsip_evsub_send_request
+- app_queue: Minor docs and logging fixes for UnpauseQueueMember.
+- app_queue: Reset all queue defaults before reload.
+- res_pjsip: Upgraded bundled pjsip to 2.13
+- doxygen: Fix doxygen errors.
+- app_signal: Add signaling applications
+- app_directory: add ability to specify configuration file
+- func_json: Enhance parsing capabilities of JSON_DECODE
+- res_stasis_snoop: Fix snoop crash
+- pbx_ael: Global variables are not expanded.
+- res_pjsip_session: Add overlap_context option.
+- app_playback.c: Fix PLAYBACKSTATUS regression.
+- res_rtp_asterisk: Don't use double math to generate timestamps
+- format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
+- res_pjsip_rfc3326: Add SIP causes support for RFC3326
+- res_rtp_asterisk: Asterisk Media Experience Score (MES)
+- Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"
+- loader: Allow declined modules to be unloaded.
+- app_broadcast: Add Broadcast application
+- func_frame_trace: Print text for text frames.
+- json.h: Add ast_json_object_real_get.
+- manager: Fix appending variables.
+- res_pjsip_transport_websocket: Add remote port to transport
+- http.c: Fix NULL pointer dereference bug
+- res_http_media_cache: Do not crash when there is no extension
+- res_rtp_asterisk: Asterisk Media Experience Score (MES)
+- pbx_app: Update outdated pbx_exec channel snapshots.
+- res_pjsip_session: Use Caller ID for extension matching.
+- res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
+- app_voicemail_odbc: Fix string overflow warning.
+- func_callerid: Warn about invalid redirecting reason.
+- res_pjsip: Fix path usage in case dialing with '@'
+- streams: Ensure that stream is closed in ast_stream_and_wait on error
+- app_sendtext: Remove references to removed applications.
+- res_geoloc: fix NULL pointer dereference bug
+- res_pjsip_aoc: Don't assume a body exists on responses.
+- app_if: Fix format truncation errors.
+- manager: AOC-S support for AOCMessage
+- res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
+- ari: Destroy body variables in channel create.
+- app_voicemail: Fix missing email in msg_create_from_file.
+- res_pjsip: Fix typo in from_domain documentation
+- res_hep: Add support for named capture agents.
+- app_if: Adds conditional branch applications
+- res_pjsip_session.c: Map empty extensions in INVITEs to s.
+- res_pjsip: Update contact_user to point out default
+- res_adsi: Fix major regression caused by media format rearchitecture.
+- res_pjsip_header_funcs: Add custom parameter support.
+- func_presencestate: Fix invalid memory access.
+- sig_analog: Fix no timeout duration.
+- xmldoc: Allow XML docs to be reloaded.
+- rtp_engine.h: Update examples using ast_format_set.
+- app_mixmonitor: Add option to use real Caller ID for voicemail.
+- pjproject: 2.13 security fixes
+- pjsip_transport_events: Fix possible use after free on transport
+- manager: prevent file access outside of config dir
+- ooh323c: not checking for IE minimum length
+- pbx_builtins: Allow Answer to return immediately.
+- chan_dahdi: Allow FXO channels to start immediately.
+- core & res_pjsip: Improve topology change handling.
+- sla: Prevent deadlock and crash due to autoservicing.
+- Build system: Avoid executable stack.
+- func_json: Fix memory leak.
+- test_json: Remove duplicated static function.
+- res_agi: Respect "transmit_silence" option for "RECORD FILE".
+- app_mixmonitor: Add option to delete files on exit.
+- manager: Update ModuleCheck documentation.
+- file.c: Don't emit warnings on winks.
+- runUnittests.sh: Save coredumps to proper directory
+- app_stack: Print proper exit location for PBXless channels.
+- chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer
+- res_pjsip: prevent crash on websocket disconnect
+- tcptls: Prevent crash when freeing OpenSSL errors.
+- res_pjsip_outbound_registration: Allow to use multiple proxies for registration
+- tests: Fix compilation errors on 32-bit.
+- res_pjsip: return all codecs on a re-INVITE without SDP
+- res_pjsip_notify: Add option support for AMI.
+- res_pjsip_logger: Add method-based logging option.
+- Dialing API: Cancel a running async thread, may not cancel all calls
+- chan_dahdi: Fix unavailable channels returning busy.
+- res_pjsip_pubsub: Prevent removing subscriptions.
+- say: Don't prepend ampersand erroneously.
+- res_crypto: handle unsafe private key files
+- audiohook: add directional awareness
+- cdr: Allow bridging and dial state changes to be ignored.
+- res_tonedetect: Add ringback support to TONE_DETECT.
+- chan_dahdi: Resolve format truncation warning.
+- res_crypto: don't modify fname in try_load_key()
+- res_crypto: use ast_file_read_dirs() to iterate
+- res_geolocation: Update wiki documentation
+- res_pjsip: Add mediasec capabilities.
+- res_prometheus: Do not crash on invisible bridges
+- res_pjsip_geolocation: Change some notices to debugs.
+- db: Fix incorrect DB tree count for AMI.
+- func_logic: Don't emit warning if both IF branches are empty.
+- features: Add no answer option to Bridge.
+- app_bridgewait: Add option to not answer channel.
+- app_amd: Add option to play audio during AMD.
+- test: initialize capture structure before freeing
+- func_export: Add EXPORT function
+- res_pjsip: Add 100rel option "peer_supported".
+- func_scramble: Fix null pointer dereference.
+- manager: be more aggressive about purging http sessions.
+- func_strings: Add trim functions.
+- res_crypto: Memory issues and uninitialized variable errors
+- res_geolocation: Fix issues exposed by compiling with -O2
+- res_crypto: don't complain about directories
+- res_pjsip: Add user=phone on From and PAID for usereqphone=yes
+- res_geolocation: Fix segfault when there's an empty element
+- res_musiconhold: Add option to not play music on hold on unanswered channels
+- res_pjsip: Add TEL URI support for basic calls.
+- res_crypto: Use EVP API's instead of legacy API's
+- test: Add coverage for res_crypto
+- res_crypto: make keys reloadable on demand for testing
+- test: Add test coverage for capture child process output
+- main/utils: allow checking for command in $PATH
+- test: Add ability to capture child process output
+- res_crypto: Don't load non-regular files in keys directory
+- func_frame_trace: Remove bogus assertion.
+- lock.c: Add AMI event for deadlocks.
+- app_confbridge: Add end_marked_any option.
+- pbx_variables: Use const char if possible.
+- res_geolocation: Add two new options to GEOLOC_PROFILE
+- res_geolocation: Allow location parameters on the profile object
+- res_geolocation: Add profile parameter suppress_empty_ca_elements
+- res_geolocation: Add built-in profiles
+- res_pjsip_sdp_rtp: Skip formats without SDP details.
+- cli: Prevent assertions on startup from bad ao2 refs.
+- pjsip: Add TLS transport reload support for certificate and key.
+- res_tonedetect: Fix typos referring to wrong variables.
+- alembic: add missing ps_endpoints columns
+- chan_dahdi.c: Resolve a format-truncation build warning.
+- res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update
+- channel.h: Remove redundant declaration.
+- features: Add transfer initiation options.
+- CI: Fixing path issue on venv check
+- CI: use Python3 virtual environment
+- general: Very minor coding guideline fixes.
+- res_geolocation: Address user issues, remove complexity, plug leaks
+- chan_iax2: Add missing options documentation.
+- app_confbridge: Fix memory leak on updated menu options.
+- Geolocation: Wiki Documentation
+- manager: Remove documentation for nonexistent action.
+- general: Improve logging levels of some log messages.
+- cdr.conf: Remove obsolete app_mysql reference.
+- general: Remove obsolete SVN references.
+- app_confbridge: Add missing AMI documentation.
+- app_meetme: Add missing AMI documentation.
+- func_srv: Document field parameter.
+- pbx_functions.c: Manually update ast_str strlen.
+- build: fix bininstall launchd issue on cross-platform build
+- db: Add AMI action to retrieve DB keys at prefix.
+- manager: Fix incomplete filtering of AMI events.
+- Update defaultbranch to 20
+- res_pjsip: delay contact pruning on Asterisk start
+- chan_dahdi: Fix buggy and missing Caller ID parameters
+- queues.conf.sample: Correction of typo
+- chan_dahdi: Add POLARITY function.
+- Makefile: Avoid git-make user conflict
+- app_confbridge: Always set minimum video update interval.
+- pbx.c: Simplify ast_context memory management.
+- geoloc_eprofile.c: Fix setting of loc_src in set_loc_src()
+- Geolocation: chan_pjsip Capability Preview
+- Geolocation: Core Capability Preview
+- general: Fix various typos.
+- cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type
+- chan_iax2: Allow compiling without OpenSSL.
+- websocket / aeap: Handle poll() interruptions better.
+- res_cliexec: Add dialplan exec CLI command.
+- features: Update documentation for automon and automixmon
+- Geolocation: Base Asterisk Prereqs
+- pbx_lua: Remove compiler warnings
+- res_prometheus: Optional load res_pjsip_outbound_registration.so
+- app_dial: Fix dial status regression.
+- db: Notify user if deleted DB entry didn't exist.
+- cli: Fix CLI blocking forever on terminating backslash
+- app_dial: Propagate outbound hook flashes.
+- res_calendar_icalendar: Send user agent in request.
+- say: Abort play loop if caller hangs up.
+- res_pjsip: allow TLS verification of wildcard cert-bearing servers
+- pbx: Add helper function to execute applications.
+- pjsip: Upgrade bundled version to pjproject 2.12.1
+- asterisk.c: Fix incompatibility warnings for remote console.
+- test_aeap_transport: disable part of failing unit test
+- sig_analog: Fix broken three-way conferencing.
+- app_voicemail: Add option to prevent message deletion.
+- res_parking: Add music on hold override option.
+- xmldocs: Improve examples.
+- res_pjsip_outbound_registration: Make max random delay configurable.
+- res_pjsip: Actually enable session timers when timers=always
+- res_pjsip_pubsub: delete scheduled notification on RLS update
+- res_pjsip_pubsub: XML sanitized RLS display name
+- app_sayunixtime: Use correct inflection for German time.
+- chan_iax2: Prevent deadlock due to duplicate autoservice.
+- loader: Prevent deadlock using tab completion.
+- res_calendar: Prevent assertion if event ends in past.
+- res_parking: Warn if out of bounds parking spot requested.
+- chan_pjsip: Only set default audio stream on hold.
+- res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI
+- res_agi: Evaluate dialplan functions and variables in agi exec if enabled
+- ast_pkgconfig.m4: AST_PKG_CONFIG_CHECK() relies on sed.
+- ari: expose channel driver's unique id to ARI channel resource
+- loader.c: Use portable printf conversion specifier for int64.
+- res_pjsip_transport_websocket: Also set the remote name.
+- res_pjsip_transport_websocket: save the original contact host
+- res_pjsip_outbound_registration: Show time until expiration
+- app_confbridge: Add function to retrieve channels.
+- chan_dahdi: Fix broken operator mode clearing.
+- GCC12: Fixes for 16+
+- GCC12: Fixes for 18+. state_id_by_topic comparing wrong value
+- core_unreal: Flip stream direction of second channel.
+- chan_dahdi: Document dial resource options.
+- chan_dahdi: Don't allow MWI FSK if channel not idle.
+- chan_dahdi: Don't append cadences on dahdi restart.
+- chan_iax2: Prevent crash if dialing RSA-only call without outkey.
+- menuselect: Don't erroneously recompile modules.
+- app_meetme: Don't erroneously set global variables.
+- asterisk.c: Warn of incompatibilities with remote console.
+- func_db: Add function to return cardinality at prefix
+- chan_dahdi: Fix insufficient array size for round robin.
+- chan_sip.c Session timers get removed on UPDATE
+- func_evalexten: Extension evaluation function.
+- file.c: Prevent formats from seeking negative offsets.
+- chan_pjsip: Add ability to send flash events.
+- cli: Add command to evaluate dialplan functions.
+- documentation: Adds versioning information.
+- samples: Remove obsolete sample configs.
+- chan_pjsip: add allow_sending_180_after_183 option
+- chan_sip: SIP route header is missing on UPDATE
+- manager: Terminate session on write error.
+- bridge_simple.c: Unhold channels on join simple bridge.
+- res_aeap & res_speech_aeap: Add Asterisk External Application Protocol
+- app_dial: Flip stream direction of outgoing channel.
+- res_pjsip_stir_shaken.c: Fix enabled when not configured.
+- res_pjsip: Always set async_operations to 1.
+- config.h: Don't use C++ keywords as argument names.
+- cdr_adaptive_odbc: Add support for SQL_DATETIME field type.
+- pjsip: Increase maximum number of format attributes.
+- AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header.
+- AST-2022-001 - res_stir_shaken/curl: Limit file size and check start.
+- func_odbc: Add SQL_ESC_BACKSLASHES dialplan function.
+- app_mf, app_sf: Return -1 if channel hangs up.
+- app_queue: Add music on hold option to Queue.
+- app_meetme: Emit warning if conference not found.
+- build: Remove obsolete leftover build references.
+- res_pjsip_header_funcs: wrong pool used tdata headers
+- deprecation cleanup: remove leftover files
+- pjproject: Update bundled to 2.12 release.
+- pbx.c: Warn if there are too many includes in a context.
+- Makefile: Disable XML doc validation
+- make_xml_documentation: Remove usage of get_sourceable_makeopts
+- chan_iax2: Fix spacing in netstats command
+- openssl: Supress deprecation warnings from OpenSSL 3.0
+- documentation: Add information on running install_prereq script in readme
+- chan_iax2: Fix perceived showing host address.
+- res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity
+- configure.ac: Use pkg-config to detect libxml2
+- time: add support for time64 libcs
+- res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request
+- app_dial: Document DIALSTATUS return values.
+- stasis_recording: Perform a complete match on requested filename.
+- download_externals: Use HTTPS for downloads
+- conversions.c: Specify that we only want to parse decimal numbers.
+- logger: workaround woefully small BUFSIZ in MUSL
+- pbx_builtins: Add missing options documentation
+- res_pjsip_pubsub: update RLS to reflect the changes to the lists
+- res_agi: Fix xmldocs bug with set music.
+- res_config_pgsql: Add text-type column check in require_pgsql()
+- app_queue: Add QueueWithdrawCaller AMI action
+- ami: Improve substring parsing for disabled events.
+- xml.c, config,c: Add stylesheets and variable list string parsing
+- xmldoc: Fix issue with xmlstarlet validation
+- core: Config and XML tweaks needed for geolocation
+- Makefile: Allow XML documentation to exist outside source files
+- build: Refactor the earlier "basebranch" commit
+- jansson: Update bundled to 2.14 version.
+- func_channel: Add lastcontext and lastexten.
+- channel.c: Clean up debug level 1.
+- configs, LICENSE: remove pbx.digium.com.
+- documentation: Add since tag to xmldocs DTD
+- asterisk: Add macro for curl user agent.
+- res_stir_shaken: refactor utility function
+- app_voicemail: Emit warning if asking for nonexistent mailbox.
+- res_pjsip_pubsub: fix Batched Notifications stop working
+- res_pjsip_pubsub: provide a display name for RLS subscriptions
+- func_db: Add validity check for key names when writing.
+- cli: Add core dump info to core show settings.
+- documentation: Adds missing default attributes.
+- app_mp3: Document and warn about HTTPS incompatibility.
+- app_mf: Add max digits option to ReceiveMF.
+- ami: Allow events to be globally disabled.
+- taskprocessor.c: Prevent crash on graceful shutdown
+- app_queue: load queues and members from Realtime when needed
+- manager.c: Simplify AMI ModuleCheck handling
+- res_prometheus.c: missing module dependency
+- res_pjsip.c: Correct minor typos in 'realm' documentation.
+- manager.c: Generate valid XML if attribute names have leading digits.
+- build_tools/make_version: Fix bashism in comparison.
+- bundled_pjproject: Add additional multipart search utils
+- chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN
+- build: Add "basebranch" to .gitreview
+- res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup
+- cdr: allow disabling CDR by default on new channels
+- res_tonedetect: Fixes some logic issues and typos
+- func_frame_drop: Fix typo referencing wrong buffer
+- res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf
+- res_http_websocket: Add a client connection timeout
+- build: Rebuild configure and autoconfig.h.in
+- sched: fix and test a double deref on delete of an executing call back
+- res_pjsip_sdp_rtp.c: Support keepalive for video streams.
+- build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD
+- main/utils: Implement ast_get_tid() for NetBSD
+- main: Enable rdtsc support on NetBSD
+- BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD
+- include: Remove unimplemented HMAC declarations
+- frame.h: Fix spelling typo
+- res_rtp_asterisk: Fix typo in flag test/set
+- bundled_pjproject: Fix srtp detection
+- res_pjsip: Make message_filter and session multipart aware
+- build: Fix issues building pjproject
+- res_pjsip: Add utils for checking media types
+- bundled_pjproject: Create generic pjsip_hdr_find functions
+- say.c: Prevent erroneous failures with 'say' family of functions.
+- documentation: Document built-in system and channel vars
+- pbx_variables: add missing ASTSBINDIR variable
+- bundled_pjproject: Make it easier to hack
+- utils.c: Remove all usages of ast_gethostbyname()
+- say.conf: fix 12pm noon logic
+- pjproject: Fix incorrect unescaping of tokens during parsing
+- app_queue.c: Support for Nordic syntax in announcements
+- dsp: Add define macro for DTMF_MATRIX_SIZE
+- ami: Add AMI event for Wink
+- cli: Add module refresh command
+- app_mp3: Throw warning on nonexistent stream
+- documentation: Add missing AMI documentation
+- tcptls.c: refactor client connection to be more robust
+- app_sf: Add full tech-agnostic SF support
+- app_queue: Fix hint updates, allow dup. hints
+- say.c: Honor requests for DTMF interruption.
+- res_pjsip_sdp_rtp: Preserve order of RTP codecs
+- bridge: Unlock channel during Local peer check.
+- test_time.c: Tolerate DST transitions
+- bundled_pjproject: Add more support for multipart bodies
+- ast_coredumper: Fix deleting results when output dir is set
+- pbx_variables: initialize uninitialized variable
+- app_queue.c: added DIALEDPEERNUMBER on outgoing channel
+- http.c: Add ability to create multiple HTTP servers
+- app.c: Throw warnings for nonexistent options
+- app_voicemail.c: Support for Danish syntax in VM
+- app_sendtext: Add ReceiveText application
+- strings: Fix enum names in comment examples
+- pbx_variables: Increase parsing capabilities of MSet
+- chan_sip: Fix crash when accessing RURI before initiating outgoing call
+- func_json: Adds JSON_DECODE function
+- configs: Updates to sample configs
+- pbx: Add variable substitution API for extensions
+- CHANGES: Correct reference to configuration file.
+- app_mf: Add full tech-agnostic MF support
+- xmldoc: Avoid whitespace around value for parameter/required.
+- progdocs: Fix Doxygen left-overs.
+- xmldoc: Correct definition for XML element 'matchInfo'.
+- progdocs: Update Makefile.
+- res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites.
+- channel: Short-circuit ast_channel_get_by_name() on empty arg.
+- res_rtp_asterisk: Addressing possible rtp range issues
+- res: Fix for Doxygen.
+- res_fax_spandsp: Add spandsp 3.0.0+ compatibility
+- main: Fix for Doxygen.
+- progdocs: Fix for Doxygen, the hidden parts.
+- progdocs: Fix grouping for latest Doxygen.
+- documentation: Standardize examples
+- config.c: Prevent UB in ast_realtime_require_field.
+- app_voicemail: Refactor email generation functions
+- stir/shaken: Avoid a compiler extension of GCC.
+- progdocs: Remove outdated references in doxyref.h.
+- logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__
+- xmldoc: Fix for Doxygen.
+- astobj2.c: Fix core when ref_log enabled
+- channels: Fix for Doxygen.
+- bridge: Deny full Local channel pair in bridge.
+- res_tonedetect: Add call progress tone detection
+- rtp_engine: Add type field for JSON RTCP Report stasis messages
+- odbc: Fix for Doxygen.
+- parking: Fix for Doxygen.
+- res_ari: Fix for Doxygen.
+- frame: Fix for Doxygen.
+- ari-stubs: Avoid 'is' as comparism with an literal.
+- BuildSystem: Consistently allow 'ye' even for Jansson.
+- stasis: Fix for Doxygen.
+- app: Fix for Doxygen.
+- res_xmpp: Fix for Doxygen.
+- channel: Fix for Doxygen.
+- chan_iax2: Fix for Doxygen.
+- res_pjsip: Fix for Doxygen.
+- bridges: Fix for Doxygen.
+- addons: Fix for Doxygen.
+- apps: Fix for Doxygen.
+- tests: Fix for Doxygen.
+- progdocs: Avoid multiple use of section labels.
+- progdocs: Use Doxygen \example correctly.
+- bridge_channel: Fix for Doxygen.
+- progdocs: Avoid 'name' with Doxygen \file.
+- app_morsecode: Fix deadlock
+- app_read: Fix custom terminator functionality regression
+- res_pjsip_callerid: Fix OLI parsing
+- build_tools: Spelling fixes
+- contrib: Spelling fixes
+- codecs: Spelling fixes
+- formats: Spelling fixes
+- CREDITS: Spelling fixes
+- addons: Spelling fixes
+- configs: Spelling fixes
+- doc: Spelling fixes
+- menuselect: Spelling fixes
+- include: Spelling fixes
+- UPGRADE.txt: Spelling fixes
+- bridges: Spelling fixes
+- apps: Spelling fixes
+- channels: Spelling fixes
+- tests: Spelling fixes
+- CHANGES: Spelling fixes
+- funcs: Spelling fixes
+- pbx: Spelling fixes
+- main: Spelling fixes
+- utils: Spelling fixes
+- Makefile: Spelling fixes
+- res: Spelling fixes
+- rest-api-templates: Spelling fixes
+- agi: Spelling fixes
+- CI: Rename 'master' node to 'built-in'
+- BuildSystem: In POSIX sh, == in place of = is undefined.
+- pbx.c: Don't remove dashes from hints on reload.
+- sig_analog: Fix truncated buffer copy
+- app_voicemail: Fix phantom voicemail bug on rerecord
+- chan_iax2: Allow both secret and outkey at dial time
+- res_snmp: As build tool, prefer pkg-config over net-snmp-config.
+- res_config_sqlite: Remove deprecated module.
+- stasis: Avoid 'dispatched' as unused variable in normal mode.
+- various: Fix GCC 11.2 compilation issues.
+- ast_coredumper: Refactor to better find things
+- strings/json: Add string delimter match, and object create with vars methods
+- STIR/SHAKEN: Option split and response codes.
+- app_queue: Add LoginTime field for member in a queue.
+- res_speech: Add a type conversion, and new engine unregister methods
+- various: Fix GCC 11 compilation issues.
+- apps/app_playback.c: Add 'mix' option to app_playback
+- BuildSystem: Check for alternate openssl packages
+- func_talkdetect.c: Fix logical errors in silence detection.
+- configure: Remove unused OpenSSL SRTP check.
+- build: prevent binary downloads for non x86 architectures
+- main/stun.c: fix crash upon STUN request timeout
+- Makefile: Use basename in a POSIX-compliant way.
+- pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+- chan_iax2: Add encryption for RSA authentication
+- res_pjsip_t38: bind UDPTL sessions like RTP
+- app_read: Fix null pointer crash
+- res_rtp_asterisk: fix memory leak
+- main/say.c: Support future dates with Q and q format params
+- res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts
+- ari: Ignore invisible bridges when listing bridges.
+- func_vmcount: Add support for multiple mailboxes
+- message.c: Support 'To' header override with AMI's MessageSend.
+- func_channel: Add CHANNEL_EXISTS function.
+- app_queue: Fix hint updates for included contexts
+- res_http_media_cache.c: Compare unaltered MIME types.
+- logger: Add custom logging capabilities
+- app_externalivr.c: Fix mixed leading whitespace in source code.
+- res_rtp_asterisk.c: Fix build failure when not building with pjproject.
+- pjproject: Add patch to fix trailing whitespace issue in rtpmap
+- app_mp3: Force output to 16 bits in mpg123
+- res_pjsip_caller_id: Add ANI2/OLI parsing
+- app_mf: Add channel agnostic MF sender
+- app_stack: Include current location if branch fails
+- test_http_media_cache.c: Fix copy/paste error during test deregistration.
+- resource_channels.c: Fix external media data option
+- func_strings: Add STRBETWEEN function
+- test_abstract_jb.c: Fix put and put_out_of_order memory leaks.
+- func_env: Add DIRNAME and BASENAME functions
+- func_sayfiles: Retrieve say file names
+- res_tonedetect: Tone detection module
+- res_snmp: Add -fPIC to _ASTCFLAGS
+- app_voicemail.c: Ability to silence instructions if greeting is present.
+- term.c: Add support for extended number format terminfo files.
+- res_srtp: Disable parsing of not enabled cryptos
+- dns.c: Load IPv6 DNS resolvers if configured.
+- bridge_softmix: Suppress error on topology change failure
+- resource_channels.c: Fix wrong external media parameter parse
+- config_options: Handle ACO arrays correctly in generated XML docs.
+- chan_iax2: Add ANI2/OLI information element
+- pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+- app_read: Allow reading # as a digit
+- res_rtp_asterisk: Automatically refresh stunaddr from DNS
+- bridge_basic: Change warning to verbose if transfer cancelled
+- app_queue: Don't reset queue stats on reload
+- res_rtp_asterisk: sqrt(.) requires the header math.h.
+- dialplan: Add one static and fix two whitespace errors.
+- sig_analog: Changes to improve electromechanical signalling compatibility
+- media_cache: Don't lock when curl the remote file
+- res_pjproject: Allow mapping to Asterisk TRACE level
+- app_milliwatt: Timing fix
+- func_math: Return integer instead of float if possible
+- app_morsecode: Add American Morse code
+- func_scramble: Audio scrambler function
+- app_originate: Add ability to set codecs
+- BuildSystem: Remove two dead exceptions for compiler Clang.
+- chan_alsa, chan_sip: Add replacement to moduleinfo
+- res_monitor: Disable building by default.
+- muted: Remove deprecated application.
+- conf2ael: Remove deprecated application.
+- res_config_sqlite: Remove deprecated module.
+- chan_vpb: Remove deprecated module.
+- chan_misdn: Remove deprecated module.
+- chan_nbs: Remove deprecated module.
+- chan_phone: Remove deprecated module.
+- chan_oss: Remove deprecated module.
+- cdr_syslog: Remove deprecated module.
+- app_dahdiras: Remove deprecated module.
+- app_nbscat: Remove deprecated module.
+- app_image: Remove deprecated module.
+- app_url: Remove deprecated module.
+- app_fax: Remove deprecated module.
+- app_ices: Remove deprecated module.
+- app_mysql: Remove deprecated module.
+- cdr_mysql: Remove deprecated module.
+- mgcp: Remove dead debug code
+- policy: Deprecate modules and add versions to others.
+- func_frame_drop: New function
+- aelparse: Accept an included context with timings.
+- format_ogg_speex: Implement a "not supported" write handler
+- cdr_adaptive_odbc: Prevent filter warnings
+- app_queue: Allow streaming multiple announcement files
+- res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern
+- res_statsd: handle non-standard meter type safely
+- app_dtmfstore: New application to store digits
+- codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother
+- res_http_media_cache: Cleanup audio format lookup in HTTP requests
+- docs: Remove embedded macro in WaitForCond XML documentation.
+- Update AMI and ARI versions for Asterisk 20.
+- AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
+- AST-2021-008 - chan_iax2: remote crash on unsupported media format
+- AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.
+- res_stasis_playback: Check for chan hangup on play_on_channels
+- res_http_media_cache.c: Fix merge errors from 18 -> master
+- res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.
+- res_http_media_cache.c: Parse media URLs to find extensions.
+- main/cdr.c: Correct Party A selection.
+- stun: Emit warning message when STUN request times out
+- app_reload: New Reload application
+- res_ari: Fix audiosocket segfault
+- res_pjsip_config_wizard.c: Add port matching support.
+- app_waitforcond: New application
+- res_stasis_playback: Send PlaybackFinish event only once for errors
+- jitterbuffer: Correct signed/unsigned mismatch causing assert
+- app_dial: Expanded A option to add caller announcement
+- core: Don't play silence for Busy() and Congestion() applications.
+- res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress
+- res_pjsip_messaging: Overwrite user in existing contact URI
+- res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter
+- pbx_builtins: Corrects SayNumber warning
+- func_lock: Add "dialplan locks show" cli command.
+- func_lock: Prevent module unloading in-use module.
+- func_lock: Fix memory corruption during unload.
+- func_lock: Fix requesters counter in error paths.
+- app_originate: Allow setting Caller ID and variables
+- menuselect: Fix description of several modules.
+- app_confbridge: New ConfKick() application
+- res_pjsip_dtmf_info: Hook flash
+- app_confbridge: New option to prevent answer supervision
+- sip_to_pjsip: Fix missing cases
+- res_pjsip_messaging: Refactor outgoing URI processing
+- func_math: Three new dialplan functions
+- STIR/SHAKEN: Add Date header, dest->tn, and URL checking.
+- res_pjsip: On partial transport reload also move factories.
+- func_volume: Add read capability to function.
+- stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing
+- res_pjsip.c: Support endpoints with domain info in username
+- res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.
+- asterisk: We've moved to Libera Chat!
+- res_rtp_asterisk: make it possible to remove SOFTWARE attribute
+- res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs
+- res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml
+- AMI: Add AMI event to expose hook flash events
+- app_voicemail: Configurable voicemail beep
+- main/file.c: Don't throw error on flash event.
+- chan_sip: Expand hook flash recognition.
+- pjsip: Add patch for resolving STUN packet lifetime issues.
+- chan_pjsip: Correct misleading trace message
+- STIR/SHAKEN: Switch to base64 URL encoding.
+- STIR/SHAKEN: OPENSSL_free serial hex from openssl.
+- STIR/SHAKEN: Fix certificate type and storage.
+- translate.c: Avoid refleak when checking for a translation path
+- res_rtp_asterisk: More robust timestamp checking
+- chan_local: Skip filtering audio formats on removed streams.
+- res_pjsip.c: OPTIONS processing can now optionally skip authentication
+- translate.c: Take sampling rate into account when checking codec's buffer size
+- svn: Switch to https scheme.
+- res_pjsip: Update documentation for the auth object
+- res_aeap: Add basic config skeleton and CLI commands.
+- bridge_channel_write_frame: Check for NULL channel
+- loader.c: Speed up deprecation metadata lookup
+- res_prometheus: Clone containers before iterating
+- loader: Output warnings for deprecated modules.
+- res_rtp_asterisk: Fix standard deviation calculation
+- res_rtp_asterisk: Don't count 0 as a minimum lost packets
+- res_rtp_asterisk: Statically declare rtp_drop_packets_data object
+- res_rtp_asterisk: Only raise flash control frame on end.
+- res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
+- res_pjsip: Give error when TLS transport configured but not supported.
+- time: Add timeval create and unit conversion functions
+- app_queue: Add alembic migration to add ringinuse to queue_members.
+- modules.conf: Fix more differing usages of assignment operators.
+- logger.conf.sample: Add more debug documentation.
+- logging: Add .log to samples and update asterisk.logrotate.
+- app_queue.c: Remove dead 'updatecdr' code.
+- queues.conf.sample: Correct 'context' documentation.
+- logger: Console sessions will now respect logger.conf dateformat= option
+- app_queue.c: Don't crash when realtime queue name is empty.
+- res_pjsip_session: Make reschedule_reinvite check for NULL topologies
+- app_queue: Only send QueueMemberStatus if status changes.
+- core_unreal: Fix deadlock with T.38 control frames.
+- res_pjsip: Add support for partial transport reload.
+- menuselect: exit non-zero in case of failure on --enable|disable options.
+- res_rtp_asterisk: Force resync on SSRC change.
+- menuselect: Add ability to set deprecated and removed versions.
+- xml: Allow deprecated_in and removed_in for MODULEINFO.
+- xml: Embed module information into core XML documentation.
+- documentation: Fix non-matching module support levels.
+- channel: Fix crash in suppress API.
+- func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds
+- app.h: Fix -Werror=zero-length-bounds compile errors in dev mode.
+- app_dial.c: Only send DTMF on first progress event.
+- res_format_attr_*: Parameter Names are Case-Insensitive.
+- chan_iax2: System Header strings is included via asterisk.h/compat.h.
+- modules.conf: Fix differing usage of assignment operators.
+- strings.h: ast_str_to_upper() and _to_lower() are not pure.
+- res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
+- res/res_rtp_asterisk: generate new SSRC on native bridge end
+- sorcery: Add support for more intelligent reloading.
+- res_pjsip_refer: Move the progress dlg release to a serializer
+- res_pjsip_registrar: Include source IP and port in log messages.
+- asterisk: Update copyright.
+- AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
+- res_format_attr_h263: Generate valid SDP fmtp for H.263+.
+- res_pjsip_nat: Don't rewrite Contact on REGISTER responses.
+- channel: Fix memory leak in suppress API.
+- res_rtp_asterisk: Check remote ICE reset and reset local ice attrb
+- pjsip: Generate progress (once) when receiving a 180 with a SDP
+- main: With Dutch language year after 2020 is not spoken in say.c
+- res_pjsip: dont return early from registration if init auth fails
+- res_fax: validate the remote/local Station ID for UTF-8 format
+- app_page.c: Don't fail to Page if beep sound file is missing
+- res_pjsip_refer: Refactor progress locking and serialization
+- res_rtp_asterisk: Add packet subtype during RTCP debug when relevant
+- res_pjsip_session: Always produce offer on re-INVITE without SDP.
+- res_odbc_transaction: correctly initialise forcecommit value from DSN.
+- res_pjsip_session.c: Check topology on re-invite.
+- res_config_pgsql: Limit realtime_pgsql() to return one (no more) record.
+- app_queue: Fix conversion of complex extension states into device states
+- app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS
+- chan_sip: Filter pass-through audio/video formats away, again.
+- func_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
+- app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute.
+- AST-2021-002: Remote crash possible when negotiating T.38
+- rtp: Enable srtp replay protection
+- res_pjsip_diversion: Fix adding more than one histinfo to Supported
+- res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
+- pjsip: Make modify_local_offer2 tolerate previous failed SDP.
+- res_pjsip_refer: Always serialize calls to refer_progress_notify
+- core_unreal: Fix T.38 faxing when using local channels.
+- format_wav: Support of MIME-type for wav16
+- chan_sip: Allow [peer] without audio (text+video).
+- chan_iax2.c: Require secret and auth method if encryption is enabled
+- app_read: Release tone zone reference on early return.
+- chan_sip: Set up calls without audio (text+video), again.
+- chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable
+- channel: Set up calls without audio (text+video), again.
+- res/res_pjsip.c: allow user=phone when number contain *#
+- chan_sip: SDP: Reject audio streams correctly.
+- main/frame: Add missing control frame names to ast_frame_subclass2str
+- res_musiconhold: Add support of various URL-schemes by MoH.
+- AC_HEADER_STDC causes a compile failure with autoconf 2.70
+- pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.
+- res_pjsip_session: Avoid sometimes-uninitialized warning with Clang.
+- res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet
+- chan_pjsip.c: Add parameters to frame in indicate.
+- Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
+- chan_sip: SDP: Sidestep stream parsing when its media is disabled.
+- chan_pjsip: Assign SIPDOMAIN after creating a channel
+- chan_pjsip: Stop queueing control frames twice on outgoing channels
+- contrib/systemd: Added note on common issues with systemd and asterisk
+- func_lock: fix multiple-channel-grant problems.
+- pbx_lua: Add LUA_VERSIONS environment variable to ./configure.
+- app_mixmonitor: cleanup datastore when monitor thread fails to launch
+- app_voicemail: Prevent deadlocks when out of ODBC database connections
+- chan_pjsip: Incorporate channel reference count into transfer_refer().
+- pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type
+- asterisk: Export additional manager functions
+- res_pjsip: Prevent segfault in UDP registration with flow transports
+- codecs: Remove test-law.
+- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
+- chan_vpb.cc: Fix compile errors.
+- res_pjsip_session.c: Fix compiler warnings.
+- res_pjsip_session: Fixed NULL active media topology handle
+- app_chanspy: Spyee information missing in ChanSpyStop AMI Event
+- res_ari: Fix wrong media uri handle for channel play
+- logger.c: Automatically add a newline to formats that don't have one
+- res_pjsip_nat.c: Create deep copies of strings when appropriate
+- res_musiconhold: Don't crash when real-time doesn't return any entries
+- res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
+- pjsip: Match lifetime of INVITE session to our session.
+- res_http_media_cache.c: Set reasonable number of redirects
+- Introduce astcachedir, to be used for temporary bucket files
+- media_cache: Fix reference leak with bucket file metadata
+- res_pjsip_stir_shaken: Fix module description
+- voicemail: add option 'e' to play greetings as early media
+- loader: Sync load- and build-time deps.
+- CHANGES: Remove already applied CHANGES update
+- res_pjsip: set Accept-Encoding to identity in OPTIONS response
+- chan_sip: Remove unused sip_socket->port.
+- bridge_basic: Fixed setup of recall channels
+- modules.conf: Align the comments for more conclusiveness.
+- app_queue: Fix deadlock between update and show queues
+- res_pjsip_outbound_registration.c: Use our own scheduler and other stuff
+- pjsip_scheduler.c: Add type ONESHOT and enhance cli show command
+- sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data
+- res_pjsip/config_transport: Load and run without OpenSSL.
+- res_stir_shaken: Include OpenSSL headers where used actually.
+- func_curl.c: Allow user to set what return codes constitute a failure.
+- AST-2020-001 - res_pjsip: Return dialog locked and referenced
+- AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
+- sip_to_pjsip.py: Handle #include globs and other fixes
+- Compiler fixes for GCC with -Og
+- Compiler fixes for GCC when printf %s is NULL
+- Compiler fixes for GCC with -Os
+- chan_sip: On authentication, pick MD5 for sure.
+- main/say: Work around gcc 9 format-truncation false positive
+- res_pjsip, res_pjsip_session: initialize local variables
+- install_prereq: Add GMime 3.0.
+- BuildSystem: Enable Lua 5.4.
+- res_pjsip_session: Restore calls to ast_sip_message_apply_transport()
+- features.conf.sample: Sample sound files incorrectly quoted
+- logger.conf.sample: add missing comment mark
+- res_pjsip: Adjust outgoing offer call pref.
+- tcptls.c: Don't close TCP client file descriptors more than once
+- resource_endpoints.c: memory leak when providing a 404 response
+- Logging: Add debug logging categories
+- pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
+- app_confbridge/bridge_softmix: Add ability to force estimated bitrate
+- app_voicemail.c: Document VMSayName interruption behavior
+- res_pjsip_sdp_rtp: Fix accidentally native bridging calls
+- res_musiconhold: Load all realtime entries, not just the first
+- channels: Don't dereference NULL pointer
+- res_pjsip_diversion: fix double 181
+- res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs
+- dsp.c: Update calls to ast_format_cmp to check result properly
+- res_pjsip_session: Fix stream name memory leak.
+- func_curl.c: Prevent crash when using CURLOPT(httpheader)
+- res_musiconhold: Start playlist after initial announcement
+- res_pjsip_session: Fix session reference leak.
+- res_stasis.c: Add compare function for bridges moh container
+- logger.h: Fix ast_trace to respect scope_level
+- chan_sip.c: Don't build by default
+- audiosocket: Fix module menuselect descriptions
+- bridge_softmix/sfu_topologies_on_join: Ignore topology change failures
+- res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined
+- res_pjsip_diversion: implement support for History-Info
+- format_cap: Perform codec lookups by pointer instead of name
+- res_pjsip_session: Fix issue with COLP and 491
+- debugging: Add enough to choke a mule
+- res_pjsip_session: Handle multi-stream re-invites better
+- realtime: Increased reg_server character size
+- res_stasis.c: Added video_single option for bridge creation
+- Bridging: Use a ref to bridge_channel's channel to prevent crash.
+- conversions: Add string to signed integer conversion functions
+- app_queue: Fix leave-empty not recording a call as abandoned
+- ast_coredumper: Fix issues with naming
+- parking: Copy parker UUID as well.
+- sip_nat_settings: Update script for latest Linux.
+- samples: Fix keep_alive_interval default in pjsip.conf.
+- chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution
+- pbx: Fix hints deadlock between reload and ExtensionState.
+- logger.c: Added a new log formatter called "plain"
+- res_speech: Bump reference on format object
+- res_pjsip_diversion: handle 181
+- app_voicemail: Process urgent messages with mailcmd
+- app_queue: Member lastpause time reseting
+- res_pjsip_session: Don't aggressively terminate on failed re-INVITE.
+- bridge_channel: Ensure text messages are zero terminated
+- res_musiconhold.c: Use ast_file_read_dir to scan MoH directory
+- scope_trace: Added debug messages and added additional macros
+- stream.c: Added 2 more debugging utils and added pos to stream string
+- chan_sip: Clear ToHost property on peer when changing to dynamic host
+- ACN: Changes specific to the core
+- Makefile: Fix certified version numbers
+- res_musiconhold.c: Prevent crash with realtime MoH
+- res_pjsip: Fix codec preference defaults.
+- vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
+- pjproject: clone sdp to protect against (nat) modifications
+- utils.c: NULL terminate ast_base64decode_string.
+- ACN: Configuration renaming for pjsip endpoint
+- res_stir_shaken: Fix memory allocation error in curl.c
+- res_pjsip_session: Ensure reused streams have correct bundle group
+- res_pjsip_registrar: Don't specify an expiration for static contacts.
+- utf8.c: Add UTF-8 validation and utility functions
+- stasis_bridge.c: Fixed wrong video_mode shown
+- vector.h: Add AST_VECTOR_SORT()
+- CI: Force publishAsteriskDocs to use python2
+- websocket / pjsip: Increase maximum packet size.
+- Prepare master for the next Asterisk version
+- acl.c: Coerce a NULL pointer into the empty string
+- pjsip: Include timer patch to prevent cancelling timer 0.
+
+### Commit Details:
+
+#### logger.h: Include SCOPE_CALL_WITH_INT_RESULT() in non-dev-mode builds.
+ Author: Sean Bright
+ Date: 2024-06-29
+
+ Fixes #785
+
+
+#### app_voicemail_odbc: Allow audio to be kept on disk
+ Author: George Joseph
+ Date: 2024-04-09
+
+ This commit adds a new voicemail.conf option 'odbc_audio_on_disk'
+ which when set causes the ODBC variant of app_voicemail to leave
+ the message and greeting audio files on disk and only store the
+ message metadata in the database. This option came from a concern
+ that the database could grow to large and cause remote access
+ and/or replication to become slow. In a clustering situation
+ with this option, all asterisk instances would share the same
+ database for the metadata and either use a shared filesystem
+ or other filesystem replication service much more suitable
+ for synchronizing files.
+
+ The changes to app_voicemail to implement this feature were actually
+ quite small but due to the complexity of the module, the actual
+ source code changes were greater. They fall into the following
+ categories:
+
+ * Tracing. The module is so complex that it was impossible to
+ figure out the path taken for various scenarios without the addition
+ of many SCOPE_ENTER, SCOPE_EXIT and ast_trace statements, even in
+ code that's not related to the functional change. Making this worse
+ was the fact that many "if" statements in this module didn't use
+ braces. Since the tracing macros add multiple statements, many "if"
+ statements had to be converted to use braces.
+
+ * Excessive use of PATH_MAX. Previous maintainers of this module
+ used PATH_MAX to allocate character arrays for filesystem paths
+ and SQL statements as though they cost nothing. In fact, PATH_MAX
+ is defined as 4096 bytes! Some functions had (and still have)
+ multiples of these. One function has 7. Given that the vast
+ majority of installations use the default spool directory path
+ `/var/spool/asterisk/voicemail`, the actual path length is usually
+ less than 80 bytes. That's over 4000 bytes wasted. It was the
+ same for SQL statement buffers. A 4K buffer for statement that
+ only needed 60 bytes. All of these PATH_MAX allocations in the
+ ODBC related code were changed to dynamically allocated buffers.
+ The rest will have to be addressed separately.
+
+ * Bug fixes. During the development of this feature, several
+ pre-existing ODBC related bugs were discovered and fixed. They
+ had to do with leaving orphaned files on disk, not preserving
+ original message ids when moving messages between folders,
+ not honoring the "formats" config parameter in certain circumstances,
+ etc.
+
+ UserNote: This commit adds a new voicemail.conf option
+ 'odbc_audio_on_disk' which when set causes the ODBC variant of
+ app_voicemail_odbc to leave the message and greeting audio files
+ on disk and only store the message metadata in the database.
+ Much more information can be found in the voicemail.conf.sample
+ file.
+
+
+#### security_agreement.c: Always add the Require and Proxy-Require headers
+ Author: George Joseph
+ Date: 2024-07-03
+
+ The `Require: mediasec` and `Proxy-Require: mediasec` headers need
+ to be sent whenever we send `Security-Client` or `Security-Verify`
+ headers but the logic to do that was only in add_security_headers()
+ in res_pjsip_outbound_register. So while we were sending them on
+ REGISTER requests, we weren't sending them on INVITE requests.
+
+ This commit moves the logic to send the two headers out of
+ res_pjsip_outbound_register:add_security_headers() and into
+ security_agreement:ast_sip_add_security_headers(). This way
+ they're always sent when we send `Security-Client` or
+ `Security-Verify`.
+
+ Resolves: #789
+
+#### stasis_channels: Use uniqueid and name to delete old snapshots
+ Author: George Joseph
+ Date: 2024-05-08
+
+ Whenver a new channel snapshot is created or when a channel is
+ destroyed, we need to delete any existing channel snapshot from
+ the snapshot cache. Historically, we used the channel->snapshot
+ pointer to delete any existing snapshots but this has two issues.
+
+ First, if something (possibly ast_channel_internal_swap_snapshots)
+ sets channel->snapshot to NULL while there's still a snapshot in
+ the cache, we wouldn't be able to delete it and it would be orphaned
+ when the channel is destroyed. Since we use the cache to list
+ channels from the CLI, AMI and ARI, it would appear as though the
+ channel was still there when it wasn't.
+
+ Second, since there are actually two caches, one indexed by the
+ channel's uniqueid, and another indexed by the channel's name,
+ deleting from the caches by pointer requires a sequential search of
+ all of the hash table buckets in BOTH caches to find the matching
+ snapshots. Not very efficient.
+
+ So, we now delete from the caches using the channel's uniqueid
+ and name. This solves both issues.
+
+ This doesn't address how channel->snapshot might have been set
+ to NULL in the first place because although we have concrete
+ evidence that it's happening, we haven't been able to reproduce it.
+
+ Resolves: #783
+
+#### xml.c: Update deprecated libxml2 API usage.
+ Author: Sean Bright
+ Date: 2024-05-23
+
+ Two functions are deprecated as of libxml2 2.12:
+
+ * xmlSubstituteEntitiesDefault
+ * xmlParseMemory
+
+ So we update those with supported API.
+
+ Additionally, `res_calendar_caldav` has been updated to use libxml2's
+ xmlreader API instead of the SAX2 API which has always felt a little
+ hacky (see deleted comment block in `res_calendar_caldav.c`).
+
+ The xmlreader API has been around since libxml2 2.5.0 which was
+ released in 2003.
+
+ Fixes #725
+
+
+#### stir_shaken: Fix memory leak, typo in config, tn canonicalization
+ Author: George Joseph
+ Date: 2024-04-25
+
+ * Fixed possible memory leak in tn_config:tn_get_etn() where we
+ weren't releasing etn if tn or eprofile were null.
+ * We now canonicalize TNs before using them for lookups or adding
+ them to Identity headers.
+ * Fixed a typo in stir_shaken.conf.sample.
+
+ Resolves: #716
+
+#### sorcery.c: Fixed crash error when executing "module reload"
+ Author: Spiridonov Dmitry
+ Date: 2024-04-14
+
+ Fixed crash error when cli "module reload". The error appears when
+ compiling with res_prometheus and using the sorcery memory cache for
+ registrations
+
+
+#### logger.h: Add SCOPE_CALL and SCOPE_CALL_WITH_RESULT
+ Author: George Joseph
+ Date: 2024-04-09
+
+ If you're tracing a large function that may call another function
+ multiple times in different circumstances, it can be difficult to
+ see from the trace output exactly which location that function
+ was called from. There's no good way to automatically determine
+ the calling location. SCOPE_CALL and SCOPE_CALL_WITH_RESULT
+ simply print out a trace line before and after the call.
+
+ The difference between SCOPE_CALL and SCOPE_CALL_WITH_RESULT is
+ that SCOPE_CALL ignores the function's return value (if any) where
+ SCOPE_CALL_WITH_RESULT allows you to specify the type of the
+ function's return value so it can be assigned to a variable.
+ SCOPE_CALL_WITH_INT_RESULT is just a wrapper for SCOPE_CALL_WITH_RESULT
+ and the "int" return type.
+
+
+#### res_stir_shaken: Fix compilation for CentOS7 (openssl 1.0.2)
+ Author: George Joseph
+ Date: 2024-04-01
+
+ * OpenSSL 1.0.2 doesn't support X509_get0_pubkey so we now use
+ X509_get_pubkey. The difference is that X509_get_pubkey requires
+ the caller to free the EVP_PKEY themselves so we now let
+ RAII_VAR do that.
+ * OpenSSL 1.0.2 doesn't support upreffing an X509_STORE so we now
+ wrap it in an ao2 object.
+ * OpenSSL 1.0.2 doesn't support X509_STORE_get0_objects to get all
+ the certs from an X509_STORE and there's no easy way to polyfill
+ it so the CLI commands that list profiles will show a "not
+ supported" message instead of listing the certs in a store.
+
+ Resolves: #676
+
+#### res_ari.c: Add additional output to ARI requests when debug is enabled
+ Author: Martin Nystroem
+ Date: 2024-03-22
+
+ When ARI debug is enabled the logs will now output http method and the uri.
+
+ Fixes: #666
+
+#### alembic: Fix compatibility with SQLAlchemy 2.0+.
+ Author: Sean Bright
+ Date: 2024-03-20
+
+ SQLAlchemy 2.0 changed the way that commits/rollbacks are handled
+ causing the final `UPDATE` to our `alembic_version_<whatever>` tables
+ to be rolled back instead of committed.
+
+ We now use one connection to determine which
+ `alembic_version_<whatever>` table to use and another to run the
+ actual migrations. This prevents the erroneous rollback.
+
+ This change is compatible with both SQLAlchemy 1.4 and 2.0.
+
+
+#### pbx_variables.c: Prevent SEGV due to stack overflow.
+ Author: Naveen Albert
+ Date: 2023-12-04
+
+ It is possible for dialplan to result in an infinite
+ recursion of variable substitution, which eventually
+ leads to stack overflow. If we detect this, abort
+ substitution and log an error for the user to fix
+ the broken dialplan.
+
+ Resolves: #480
+
+ UpgradeNote: The maximum amount of dialplan recursion
+ using variable substitution (such as by using EVAL_EXTEN)
+ is capped at 15.
+
+
+#### res_pjsip: Fix alembic downgrade for boolean columns.
+ Author: Sean Bright
+ Date: 2024-03-18
+
+ When downgrading, ensure that we don't touch columns that didn't
+ actually change during upgrade.
+
+
+#### alembic: Quote new MySQL keyword 'qualify.'
+ Author: Sean Bright
+ Date: 2024-03-15
+
+ Fixes #651
+
+
+#### asterisk.c: Fix sending incorrect messages to systemd notify
+ Author: Ivan Poddubny
+ Date: 2024-05-05
+
+ Send "RELOADING=1" instead of "RELOAD=1" to follow the format
+ expected by systemd (see sd_notify(3) man page).
+
+ Do not send STOPPING=1 in remote console mode:
+ attempting to execute "asterisk -rx" by the main process leads to
+ a warning if NotifyAccess=main (the default) or to a forced termination
+ if NotifyAccess=all.
+
+
+#### tcptls/iostream: Add support for setting SNI on client TLS connections
+ Author: George Joseph
+ Date: 2024-04-23
+
+ If the hostname field of the ast_tcptls_session_args structure is
+ set (which it is for websocket client connections), that hostname
+ will now automatically be used in an SNI TLS extension in the client
+ hello.
+
+ Resolves: #713
+
+ UserNote: Secure websocket client connections now send SNI in
+ the TLS client hello.
+
+
+#### make_buildopts_h: Always include DETECT_DEADLOCKS
+ Author: George Joseph
+ Date: 2024-04-27
+
+ Since DETECT_DEADLOCKS is now split from DEBUG_THREADS, it must
+ always be included in buildopts.h instead of only when
+ ADD_CFLAGS_TO_BUILDOPTS_H is defined. A SEGV will result otherwise.
+
+ Resolves: #719
+
+#### alembic: Correct NULLability of PJSIP id columns.
+ Author: Sean Bright
+ Date: 2024-04-06
+
+ Fixes #695
+
+
+#### rtp_engine and stun: call ast_register_atexit instead of ast_register_cleanup
+ Author: George Joseph
+ Date: 2024-04-02
+
+ rtp_engine.c and stun.c were calling ast_register_cleanup which
+ is skipped if any loadable module can't be cleanly unloaded
+ when asterisk shuts down. Since this will always be the case,
+ their cleanup functions never get run. In a practical sense
+ this makes no difference since asterisk is shutting down but if
+ you're in development mode and trying to use the leak sanitizer,
+ the leaks from both of those modules clutter up the output.
+
+
+#### manager.c: Add missing parameters to Login documentation
+ Author: George Joseph
+ Date: 2024-04-03
+
+ * Added the AuthType and Key parameters for MD5 authentication.
+
+ * Added the Events parameter.
+
+ Resolves: #689
+
+#### Fix incorrect application and function documentation references
+ Author: George Joseph
+ Date: 2024-04-01
+
+ There were a few references in the embedded documentation XML
+ where the case didn't match or where the referenced app or function
+ simply didn't exist any more. These were causing 404 responses
+ in docs.asterisk.org.
+
+
+#### Initial commit for certified-20.7
+ Author: George Joseph
+ Date: 2024-03-18
+
+
+#### res_pjsip_stir_shaken.c: Add checks for missing parameters
+ Author: George Joseph
+ Date: 2024-03-11
+
+ * Added checks for missing session, session->channel and rdata
+ in stir_shaken_incoming_request.
+
+ * Added checks for missing session, session->channel and tdata
+ in stir_shaken_outgoing_request.
+
+ Resolves: #645
+
+#### app_dial: Add dial time for progress/ringing.
+ Author: Naveen Albert
+ Date: 2024-02-08
+
+ Add a timeout option to control the amount of time
+ to wait if no early media is received before giving
+ up. This allows aborting early if the destination
+ is not being responsive.
+
+ Resolves: #588
+
+ UserNote: The timeout argument to Dial now allows
+ specifying the maximum amount of time to dial if
+ early media is not received.
+
+
+#### app_voicemail: Properly reinitialize config after unit tests.
+ Author: Naveen Albert
+ Date: 2024-02-29
+
+ Most app_voicemail unit tests were not properly cleaning up
+ after themselves after running. This led to test mailboxes
+ lingering around in the system. It also meant that if any
+ unit tests in app_voicemail that create mailboxes were executed
+ and the module was not unloaded/loaded again prior to running
+ the test_voicemail_vm_info unit test, Asterisk would segfault
+ due to an attempt to copy a NULL string.
+
+ The load_config test did actually have logic to reinitialize
+ the config after the test. However, this did not work in practice
+ since load_config() would not reload the config since voicemail.conf
+ had not changed during the test; thus, additional logic has been
+ added to ensure that voicemail.conf is truly reloaded, after any
+ unit tests which modify the users list.
+
+ This prevents the SEGV due to invalid mailboxes lingering around,
+ and also ensures that the system state is restored to what it was
+ prior to the tests running.
+
+ Resolves: #629
+
+#### app_queue.c : fix "queue add member" usage string
+ Author: Shaaah
+ Date: 2024-01-23
+
+ Fixing bracket placement in the "queue add member" cli usage string.
+
+
+#### app_voicemail: Allow preventing mark messages as urgent.
+ Author: Naveen Albert
+ Date: 2024-02-24
+
+ This adds an option to allow preventing callers from leaving
+ messages marked as 'urgent'.
+
+ Resolves: #619
+
+ UserNote: The leaveurgent mailbox option can now be used to
+ control whether callers may leave messages marked as 'Urgent'.
+
+
+#### res_pjsip: Use consistent type for boolean columns.
+ Author: Sean Bright
+ Date: 2024-02-27
+
+ This migrates the relevant schema objects from the `('yes', 'no')`
+ definition to the `('0', '1', 'off', 'on', 'false', 'true', 'yes', 'no')`
+ one.
+
+ Fixes #617
+
+
+#### attestation_config.c: Use ast_free instead of ast_std_free
+ Author: George Joseph
+ Date: 2024-03-05
+
+ In as_check_common_config, we were calling ast_std_free on
+ raw_key but raw_key was allocated with ast_malloc so it
+ should be freed with ast_free.
+
+ Resolves: #636
+
+#### Makefile: Add stir_shaken/cache to directories created on install
+ Author: George Joseph
+ Date: 2024-03-04
+
+ The default location for the stir_shaken cache is
+ /var/lib/asterisk/keys/stir_shaken/cache but we were only creating
+ /var/lib/asterisk/keys/stir_shaken on istall. We now create
+ the cache sub-directory.
+
+ Resolves: #634
+
+#### Stir/Shaken Refactor
+ Author: George Joseph
+ Date: 2023-10-26
+
+ Why do we need a refactor?
+
+ The original stir/shaken implementation was started over 3 years ago
+ when little was understood about practical implementation. The
+ result was an implementation that wouldn't actually interoperate
+ with any other stir-shaken implementations.
+
+ There were also a number of stir-shaken features and RFC
+ requirements that were never implemented such as TNAuthList
+ certificate validation, sending Reason headers in SIP responses
+ when verification failed but we wished to continue the call, and
+ the ability to send Media Key(mky) grants in the Identity header
+ when the call involved DTLS.
+
+ Finally, there were some performance concerns around outgoing
+ calls and selection of the correct certificate and private key.
+ The configuration was keyed by an arbitrary name which meant that
+ for every outgoing call, we had to scan the entire list of
+ configured TNs to find the correct cert to use. With only a few
+ TNs configured, this wasn't an issue but if you have a thousand,
+ it could be.
+
+ What's changed?
+
+ * Configuration objects have been refactored to be clearer about
+ their uses and to fix issues.
+ * The "general" object was renamed to "verification" since it
+ contains parameters specific to the incoming verification
+ process. It also never handled ca_path and crl_path
+ correctly.
+ * A new "attestation" object was added that controls the
+ outgoing attestation process. It sets default certificates,
+ keys, etc.
+ * The "certificate" object was renamed to "tn" and had it's key
+ change to telephone number since outgoing call attestation
+ needs to look up certificates by telephone number.
+ * The "profile" object had more parameters added to it that can
+ override default parameters specified in the "attestation"
+ and "verification" objects.
+ * The "store" object was removed altogther as it was never
+ implemented.
+
+ * We now use libjwt to create outgoing Identity headers and to
+ parse and validate signatures on incoming Identiy headers. Our
+ previous custom implementation was much of the source of the
+ interoperability issues.
+
+ * General code cleanup and refactor.
+ * Moved things to better places.
+ * Separated some of the complex functions to smaller ones.
+ * Using context objects rather than passing tons of parameters
+ in function calls.
+ * Removed some complexity and unneeded encapsuation from the
+ config objects.
+
+ Resolves: #351
+ Resolves: #46
+
+ UserNote: Asterisk's stir-shaken feature has been refactored to
+ correct interoperability, RFC compliance, and performance issues.
+ See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more
+ information.
+
+ UpgradeNote: The stir-shaken refactor is a breaking change but since
+ it's not working now we don't think it matters. The
+ stir_shaken.conf file has changed significantly which means that
+ existing ones WILL need to be changed. The stir_shaken.conf.sample
+ file in configs/samples/ has quite a bit more information. This is
+ also an ABI breaking change since some of the existing objects
+ needed to be changed or removed, and new ones added. Additionally,
+ if res_stir_shaken is enabled in menuselect, you'll need to either
+ have the development package for libjwt v1.15.3 installed or use
+ the --with-libjwt-bundled option with ./configure.
+
+
+#### alembic: Synchronize alembic heads between supported branches.
+ Author: Sean Bright
+ Date: 2024-02-28
+
+ This adds a dummy migration to 18 and 20 so that our alembic heads are
+ synchronized across all supported branches.
+
+ In this case the migration we are stubbing (24c12d8e9014) is:
+
+ https://github.com/asterisk/asterisk/commit/775352ee6c2a5bcd4f0e3df51aee5d1b0abf4cbe
+
+#### translate.c: implement new direct comp table mode
+ Author: Sebastian Jennen
+ Date: 2024-02-25
+
+ The new mode lists for each codec translation the actual real cost in cpu microseconds per second translated audio.
+ This allows to compare the real cpu usage of translations and helps in evaluation of codec implementation changes regarding performance (regression testing).
+
+ - add new table mode
+ - hide the 999999 comp values, as these only indicate an issue with transcoding
+ - hide the 0 values, as these also do not contain any information (only indicate a multistep transcoding)
+
+ Resolves: #601
+
+#### README.md: Removed outdated link
+ Author: Shyju Kanaprath
+ Date: 2024-02-23
+
+ Removed outdated link http://www.quicknet.net from README.md
+
+ cherry-pick-to: 18
+ cherry-pick-to: 20
+ cherry-pick-to: 21
+
+#### strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
+ Author: Sean Bright
+ Date: 2024-02-17
+
+ If a dynamic string is created with an initial length of 0,
+ `ast_str_buffer(…)` will return an invalid pointer.
+
+ This was a secondary discovery when fixing #65.
+
+
+#### res_rtp_asterisk.c: Correct coefficient in MOS calculation.
+ Author: romryz
+ Date: 2024-02-06
+
+ Media Experience Score relies on incorrect pseudo_mos variable
+ calculation. According to forming an opinion section of the
+ documentation, calculation relies on ITU-T G.107 standard:
+
+ https://docs.asterisk.org/Deployment/Media-Experience-Score/#forming-an-opinion
+
+ ITU-T G.107 Annex B suggests to calculate MOS with a coefficient
+ "seven times ten to the power of negative six", 7 * 10^(-6). which
+ would mean 6 digits after the decimal point. Current implementation
+ has 7 digits after the decimal point, which downrates the calls.
+
+ Fixes: #597
+
+#### dsp.c: Fix and improve potentially inaccurate log message.
+ Author: Naveen Albert
+ Date: 2024-02-09
+
+ If ast_dsp_process is called with a codec besides slin, ulaw,
+ or alaw, a warning is logged that in-band DTMF is not supported,
+ but this message is not always appropriate or correct, because
+ ast_dsp_process is much more generic than just DTMF detection.
+
+ This logs a more generic message in those cases, and also improves
+ codec-mismatch logging throughout dsp.c by ensuring incompatible
+ codecs are printed out.
+
+ Resolves: #595
+
+#### pjsip show channelstats: Prevent possible segfault when faxing
+ Author: George Joseph
+ Date: 2024-02-09
+
+ Under rare circumstances, it's possible for the original audio
+ session in the active_media_state default_session to be corrupted
+ instead of removed when switching to the t38/image media session
+ during fax negotiation. This can cause a segfault when a "pjsip
+ show channelstats" attempts to print that audio media session's
+ rtp statistics. In these cases, the active_media_state
+ topology is correctly showing only a single t38/image stream
+ so we now check that there's an audio stream in the topology
+ before attempting to use the audio media session to get the rtp
+ statistics.
+
+ Resolves: #592
+
+#### Reduce startup/shutdown verbose logging
+ Author: George Joseph
+ Date: 2024-01-31
+
+ When started with a verbose level of 3, asterisk can emit over 1500
+ verbose message that serve no real purpose other than to fill up
+ logs. When asterisk shuts down, it emits another 1100 that are of
+ even less use. Since the testsuite runs asterisk with a verbose
+ level of 3, and asterisk starts and stops for every one of the 700+
+ tests, the number of log messages is staggering. Besides taking up
+ resources, it also makes it hard to debug failing tests.
+
+ This commit changes the log level for those verbose messages to 5
+ instead of 3 which reduces the number of log messages to only a
+ handful. Of course, NOTICE, WARNING and ERROR message are
+ unaffected.
+
+ There's also one other minor change...
+ ast_context_remove_extension_callerid2() logs a DEBUG message
+ instead of an ERROR if the extension you're deleting doesn't exist.
+ The pjsip_config_wizard calls that function to clean up the config
+ and has been triggering that annoying error message for years.
+
+ Resolves: #582
+
+#### configure: Rerun bootstrap on modern platform.
+ Author: Naveen Albert
+ Date: 2024-02-12
+
+ The last time configure was run, it was run on a system that
+ did not enable -std=gnu11 by default, which meant that the
+ restrict qualifier would not be recognized on certain platforms.
+ This regenerates the configure files from running bootstrap.sh,
+ so that these should be recognized on all supported platforms.
+
+ Resolves: #586
+
+#### Upgrade bundled pjproject to 2.14.
+ Author: Ben Ford
+ Date: 2024-02-05
+
+ Fixes: #406
+
+ UserNote: Bundled pjproject has been upgraded to 2.14. For more
+ information on what all is included in this change, check out the
+ pjproject Github page: https://github.com/pjsip/pjproject/releases
+
+
+#### app_speech_utils.c: Allow partial speech results.
+ Author: cmaj
+ Date: 2024-02-02
+
+ Adds 'p' option to SpeechBackground() application.
+ With this option, when the app timeout is reached,
+ whatever the backend speech engine collected will
+ be returned as if it were the final, full result.
+ (This works for engines that make partial results.)
+
+ Resolves: #572
+
+ UserNote: The SpeechBackground dialplan application now supports a 'p'
+ option that will return partial results from speech engines that
+ provide them when a timeout occurs.
+
+
+#### utils: Make behavior of ast_strsep* match strsep.
+ Author: Joshua C. Colp
+ Date: 2024-01-31
+
+ Given the scenario of passing an empty string to the
+ ast_strsep functions the functions would return NULL
+ instead of an empty string. This is counter to how
+ strsep itself works.
+
+ This change alters the behavior of the functions to
+ match that of strsep.
+
+ Fixes: #565
+
+#### app_chanspy: Add 'D' option for dual-channel audio
+ Author: Mike Bradeen
+ Date: 2024-01-31
+
+ Adds the 'D' option to app chanspy that causes the input and output
+ frames of the spied channel to be interleaved in the spy output frame.
+ This allows the input and output of the spied channel to be decoded
+ separately by the receiver.
+
+ If the 'o' option is also set, the 'D' option is ignored as the
+ audio being spied is inherently one direction.
+
+ Fixes: #569
+
+ UserNote: The ChanSpy application now accepts the 'D' option which
+ will interleave the spied audio within the outgoing frames. The
+ purpose of this is to allow the audio to be read as a Dual channel
+ stream with separate incoming and outgoing audio. Setting both the
+ 'o' option and the 'D' option and results in the 'D' option being
+ ignored.
+
+
+#### app_if: Fix next priority calculation.
+ Author: Naveen Albert
+ Date: 2024-01-28
+
+ Commit fa3922a4d28860d415614347d9f06c233d2beb07 fixed
+ a branching issue but "overshoots" when calculating
+ the next priority. This fixes that; accompanying
+ test suite tests have also been extended.
+
+ Resolves: #560
+
+#### res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
+ Author: Sean Bright
+ Date: 2024-01-29
+
+ The existing code prevented IPv6 addresses from being properly parsed.
+
+ Fixes #558
+
+
+#### BuildSystem: Bump autotools versions on OpenBSD.
+ Author: Brad Smith
+ Date: 2024-01-27
+
+ Bump up to the more commonly used and modern versions of
+ autoconf and automake.
+
+
+#### main/utils: Simplify the FreeBSD ast_get_tid() handling
+ Author: Brad Smith
+ Date: 2024-01-27
+
+ FreeBSD has had kernel threads for 20+ years.
+
+
+#### res_pjsip_session.c: Correctly format SDP connection addresses.
+ Author: Sean Bright
+ Date: 2024-01-27
+
+ Resolves a regression identified by @justinludwig involving the
+ rendering of IPv6 addresses in outgoing SDP.
+
+ Also updates `media_address` on PJSIP endpoints so that if we are able
+ to parse the configured value as an IP we store it in a format that we
+ can directly use later. Based on my reading of the code it appeared
+ that one could configure `media_address` as:
+
+ ```
+ [foo]
+ type = endpoint
+ ...
+ media_address = [2001:db8::]
+ ```
+
+ And that value would be blindly copied into the outgoing SDP without
+ regard to its format.
+
+ Fixes #541
+
+
+#### rtp_engine.c: Correct sample rate typo for L16/44100.
+ Author: Sean Bright
+ Date: 2024-01-28
+
+ Fixes #555
+
+
+#### manager.c: Fix erroneous reloads in UpdateConfig.
+ Author: Naveen Albert
+ Date: 2024-01-25
+
+ Currently, a reload will always occur if the
+ Reload header is provided for the UpdateConfig
+ action. However, we should not be doing a reload
+ if the header value has a falsy value, per the
+ documentation, so this makes the reload behavior
+ consistent with the existing documentation.
+
+ Resolves: #551
+
+#### res_calendar_icalendar: Print iCalendar error on parsing failure.
+ Author: Naveen Albert
+ Date: 2023-12-14
+
+ If libical fails to parse a calendar, print the error message it provdes.
+
+ Resolves: #492
+
+#### app_confbridge: Don't emit warnings on valid configurations.
+ Author: Sean Bright
+ Date: 2024-01-21
+
+ The numeric bridge profile options `internal_sample_rate` and
+ `maximum_sample_rate` are documented to accept the special values
+ `auto` and `none`, respectively. While these values currently work,
+ they also emit warnings when used which could be confusing for users.
+
+ In passing, also ensure that we only accept the documented range of
+ sample rate values between 8000 and 192000.
+
+ Fixes #546
+
+
+#### app_voicemail: add NoOp alembic script to maintain sync
+ Author: Mike Bradeen
+ Date: 2024-01-17
+
+ Adding a NoOp alembic script for the voicemail database to maintain
+ version sync with other branches.
+
+ Fixes: #527
+
+#### chan_dahdi: Allow MWI to be manually toggled on channels.
+ Author: Naveen Albert
+ Date: 2023-11-10
+
+ This adds a CLI command to manually toggle the MWI status
+ of a channel, useful for troubleshooting or resetting
+ MWI devices, similar to the capabilities offered with
+ SIP messaging to manually control MWI status.
+
+ UserNote: The 'dahdi set mwi' now allows MWI on channels
+ to be manually toggled if needed for troubleshooting.
+
+ Resolves: #440
+
+#### chan_rtp.c: MulticastRTP missing refcount without codec option
+ Author: PeterHolik
+ Date: 2024-01-15
+
+ Fixes: #529
+
+#### chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
+ Author: PeterHolik
+ Date: 2024-01-16
+
+ Fixes: asterisk#536
+
+#### func_frame_trace: Add CLI command to dump frame queue.
+ Author: Naveen Albert
+ Date: 2024-01-12
+
+ This adds a simple CLI command that can be used for
+ analyzing all frames currently queued to a channel.
+
+ A couple log messages are also adjusted to be more
+ useful in tracing bridging problems.
+
+ Resolves: #533
+
+#### logger: Fix linking regression.
+ Author: Naveen Albert
+ Date: 2024-01-16
+
+ Commit 008731b0a4b96c4e6c340fff738cc12364985b64
+ caused a regression by resulting in logger.xml
+ being compiled and linked into the asterisk
+ binary in lieu of logger.c on certain platforms
+ if Asterisk was compiled in dev mode.
+
+ To fix this, we ensure the file has a unique
+ name without the extension. Most existing .xml
+ files have been named differently from any
+ .c files in the same directory or did not
+ pose this issue.
+
+ channels/pjsip/dialplan_functions.xml does not
+ pose this issue but is also being renamed
+ to adhere to this policy.
+
+ Resolves: #539
+
+#### Revert "core & res_pjsip: Improve topology change handling."
+ Author: George Joseph
+ Date: 2024-01-12
+
+ This reverts commit 315eb551dbd18ecd424a2f32179d4c1f6f6edd26.
+
+ Over the past year, we've had several reports of "topology storms"
+ occurring where 2 external facing channels connected by one or more
+ local channels and bridges will get themselves in a state where
+ they continually send each other topology change requests. This
+ usually manifests itself in no-audio calls and a flood of
+ "Exceptionally long queue length" messages. It appears that this
+ commit is the cause so we're reverting it for now until we can
+ determine a more appropriate solution.
+
+ Resolves: #530
+
+#### menuselect: Use more specific error message.
+ Author: Naveen Albert
+ Date: 2024-01-04
+
+ Instead of using the same error message for
+ missing dependencies and conflicts, be specific
+ about what actually went wrong.
+
+ Resolves: #520
+
+#### res_pjsip_nat: Fix potential use of uninitialized transport details
+ Author: Maximilian Fridrich
+ Date: 2024-01-08
+
+ The ast_sip_request_transport_details must be zero initialized,
+ otherwise this could lead to a SEGV.
+
+ Resolves: #509
+
+#### app_if: Fix faulty EndIf branching.
+ Author: Naveen Albert
+ Date: 2023-12-23
+
+ This fixes faulty branching logic for the
+ EndIf application. Instead of computing
+ the next priority, which should be done
+ for false conditionals or ExitIf, we should
+ simply advance to the next priority.
+
+ Resolves: #341
+
+#### manager.c: Fix regression due to using wrong free function.
+ Author: Naveen Albert
+ Date: 2023-12-26
+
+ Commit 424be345639d75c6cb7d0bd2da5f0f407dbd0bd5 introduced
+ a regression by calling ast_free on memory allocated by
+ realpath. This causes Asterisk to abort when executing this
+ function. Since the memory is allocated by glibc, it should
+ be freed using ast_std_free.
+
+ Resolves: #513
+
+#### config_options.c: Fix truncation of option descriptions.
+ Author: Naveen Albert
+ Date: 2023-11-09
+
+ This increases the format width of option descriptions
+ to avoid needless truncation for longer descriptions.
+
+ Resolves: #428
+
+#### manager.c: Improve clarity of "manager show connected".
+ Author: Naveen Albert
+ Date: 2023-12-05
+
+ Improve the "manager show connected" CLI command
+ to clarify that the last two columns are permissions
+ related, not counts, and use sufficient widths
+ to consistently display these values.
+
+ ASTERISK-30143 #close
+ Resolves: #482
+
+
+#### make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
+ Author: Sean Bright
+ Date: 2023-12-01
+
+ Although `make_xml_documentation`'s `print_dependencies` command was
+ corrected by the previous fix (#461) for #142, the `create_xml` was
+ not properly handling `LOCAL_MOD_SUBDIRS` XML documentation.
+
+
+#### general: Fix broken links.
+ Author: Naveen Albert
+ Date: 2023-11-09
+
+ This fixes a number of broken links throughout the
+ tree, mostly caused by wiki.asterisk.org being replaced
+ with docs.asterisk.org, which should eliminate the
+ need for sporadic fixes as in f28047db36a70e81fe373a3d19132c43adf3f74b.
+
+ Resolves: #430
+
+#### MergeApproved.yml: Remove unneeded concurrency
+ Author: George Joseph
+ Date: 2023-12-06
+
+ The concurrency parameter on the MergeAndCherryPick job has
+ been rmeoved. It was a hold-over from earlier days.
+
+
+#### app_dial: Add option "j" to preserve initial stream topology of caller
+ Author: Maximilian Fridrich
+ Date: 2023-11-30
+
+ Resolves: #462
+
+ UserNote: The option "j" is now available for the Dial application which
+ uses the initial stream topology of the caller to create the outgoing
+ channels.
+
+
+#### ast_coredumper: Increase reliability
+ Author: George Joseph
+ Date: 2023-11-11
+
+ Instead of searching for the asterisk binary and the modules in the
+ filesystem, we now get their locations, along with libdir, from
+ the coredump itself...
+
+ For the binary, we can use `gdb -c <coredump> ... "info proc exe"`.
+ gdb can print this even without having the executable and symbols.
+
+ Once we have the binary, we can get the location of the modules with
+ `gdb ... "print ast_config_AST_MODULE_DIR`
+
+ If there was no result then either it's not an asterisk coredump
+ or there were no symbols loaded. Either way, it's not usable.
+
+ For libdir, we now run "strings" on the note0 section of the
+ coredump (which has the shared library -> memory address xref) and
+ search for "libasteriskssl|libasteriskpj", then take the dirname.
+
+ Since we're now getting everything from the coredump, it has to be
+ correct as long as we're not crossing namespace boundaries like
+ running asterisk in a docker container but trying to run
+ ast_coredumper from the host using a shared file system (which you
+ shouldn't be doing).
+
+ There is still a case for using --asterisk-bin and/or --libdir: If
+ you've updated asterisk since the coredump was taken, the binary,
+ libraries and modules won't match the coredump which will render it
+ useless. If you can restore or rebuild the original files that
+ match the coredump and place them in a temporary directory, you can
+ use --asterisk-bin, --libdir, and a new --moddir option to point to
+ them and they'll be correctly captured in a tarball created
+ with --tarball-coredumps. If you also use --tarball-config, you can
+ use a new --etcdir option to point to what normally would be the
+ /etc/asterisk directory.
+
+ Also addressed many "shellcheck" findings.
+
+ Resolves: #445
+
+#### logger.c: Move LOG_GROUP documentation to dedicated XML file.
+ Author: Sean Bright
+ Date: 2023-12-01
+
+ The `get_documentation` awk script will only extract the first
+ DOCUMENTATION block that it finds in a given file. This is by design
+ (9bc2127) to prevent AMI event documentation from being pulled in to
+ the core.xml documentation file.
+
+ Because of this, the `LOG_GROUP` documentation added in 89709e2 was
+ not being properly extracted and was missing fom the resulting XML
+ documentation file. This commit moves the `LOG_GROUP` documentation to
+ a separate `logger.xml` file.
+
+
+#### res_odbc.c: Allow concurrent access to request odbc connections
+ Author: Matthew Fredrickson
+ Date: 2023-11-30
+
+ There are valid scenarios where res_odbc's connection pool might have some dead
+ or stuck connections while others are healthy (imagine network
+ elements/firewalls/routers silently timing out connections to a single DB and a
+ single IP address, or a heterogeneous connection pool connected to potentially
+ multiple IPs/instances of a replicated DB using a DNS front end for load
+ balancing and one replica fails).
+
+ In order to time out those unhealthy connections without blocking access to
+ other parts of Asterisk that may attempt access to the connection pool, it would
+ be beneficial to not lock/block access around the entire pool in
+ _ast_odbc_request_obj2 while doing potentially blocking operations on connection
+ pool objects such as the connection_dead() test, odbc_obj_connect(), or by
+ dereferencing a struct odbc_obj for the last time and triggering a
+ odbc_obj_disconnect().
+
+ This would facilitate much quicker and concurrent timeout of dead connections
+ via the connection_dead() test, which could block potentially for a long period
+ of time depending on odbc.ini or other odbc connector specific timeout settings.
+
+ This also would make rapid failover (in the clustered DB scenario) much quicker.
+
+ This patch changes the locking in _ast_odbc_request_obj2() to not lock around
+ odbc_obj_connect(), _disconnect(), and connection_dead(), while continuing to
+ lock around truly shared, non-immutable state like the connection_cnt member and
+ the connections list on struct odbc_class.
+
+ Fixes: #465
+
+#### res_pjsip_header_funcs.c: Check URI parameter length before copying.
+ Author: Sean Bright
+ Date: 2023-12-04
+
+ Fixes #477
+
+
+#### config.c: Log #exec include failures.
+ Author: Sean Bright
+ Date: 2023-11-22
+
+ If the script referenced by `#exec` does not exist, writes anything to
+ stderr, or exits abnormally or with a non-zero exit status, we log
+ that to Asterisk's error logging channel.
+
+ Additionally, write out a warning if the script produces no output.
+
+ Fixes #259
+
+
+#### make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
+ Author: Sean Bright
+ Date: 2023-11-27
+
+ If LOCAL_MOD_SUBDIRS contains absolute paths, do not prefix them with
+ the path to Asterisk's source tree.
+
+ Fixes #142
+
+
+#### app_voicemail.c: Completely resequence mailbox folders.
+ Author: Sean Bright
+ Date: 2023-11-27
+
+ Resequencing is a process that occurs when we open a voicemail folder
+ and discover that there are gaps between messages (e.g. `msg0000.txt`
+ is missing but `msg0001.txt` exists). Resequencing involves shifting
+ the existing messages down so we end up with a sequential list of
+ messages.
+
+ Currently, this process stops after reaching a threshold based on the
+ message limit (`maxmsg`) configured on the current folder. However, if
+ `maxmsg` is lowered when a voicemail folder contains more than
+ `maxmsg + 10` messages, resequencing will not run completely leaving
+ the mailbox in an inconsistent state.
+
+ We now resequence up to the maximum number of messages permitted by
+ `app_voicemail` (currently hard-coded at 9999 messages).
+
+ Fixes #86
+
+
+#### sig_analog: Fix channel leak when mwimonitor is enabled.
+ Author: Naveen Albert
+ Date: 2023-11-24
+
+ When mwimonitor=yes is enabled for an FXO port,
+ the do_monitor thread will launch mwi_thread if it thinks
+ there could be MWI on an FXO channel, due to the noise
+ threshold being satisfied. This, in turns, calls
+ analog_ss_thread_start in sig_analog. However, unlike
+ all other instances where __analog_ss_thread is called
+ in sig_analog, this call path does not properly set
+ pvt->ss_astchan to the Asterisk channel, which means
+ that the Asterisk channel is NULL when __analog_ss_thread
+ starts executing. As a result, the thread exits and the
+ channel is never properly cleaned up by calling ast_hangup.
+
+ This caused issues with do_monitor on incoming calls,
+ as it would think the channel was still owned even while
+ receiving events, leading to an infinite barrage of
+ warning messages; additionally, the channel would persist
+ improperly.
+
+ To fix this, the assignment is added to the call path
+ where it is missing (which is only used for mwi_thread).
+ A warning message is also added since previously there
+ was no indication that __analog_ss_thread was exiting
+ abnormally. This resolves both the channel leak and the
+ condition that led to the warning messages.
+
+ Resolves: #458
+
+#### res_rtp_asterisk.c: Update for OpenSSL 3+.
+ Author: Sean Bright
+ Date: 2023-11-20
+
+ In 5ac5c2b0 we defined `OPENSSL_SUPPRESS_DEPRECATED` to silence
+ deprecation warnings. This commit switches over to using
+ non-deprecated API.
+
+
+#### alembic: Update list of TLS methods available on ps_transports.
+ Author: Sean Bright
+ Date: 2023-11-14
+
+ Related to #221 and #222.
+
+ Also adds `*.ini` to the `.gitignore` file in ast-db-manage for
+ convenience.
+
+
+#### func_channel: Expose previously unsettable options.
+ Author: Naveen Albert
+ Date: 2023-11-11
+
+ Certain channel options are not set anywhere or
+ exposed in any way to users, making them unusable.
+ This exposes some of these options which make sense
+ for users to manipulate at runtime.
+
+ Resolves: #442
+
+#### app.c: Allow ampersands in playback lists to be escaped.
+ Author: Sean Bright
+ Date: 2023-11-07
+
+ Any function or application that accepts a `&`-separated list of
+ filenames can now include a literal `&` in a filename by wrapping the
+ entire filename in single quotes, e.g.:
+
+ ```
+ exten = _X.,n,Playback('https://example.com/sound.cgi?a=b&c=d'&hello-world)
+ ```
+
+ Fixes #172
+
+ UpgradeNote: Ampersands in URLs passed to the `Playback()`,
+ `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or
+ `Queue()` applications as filename arguments can now be escaped by
+ single quoting the filename. Additionally, this is also possible when
+ using the `CONFBRIDGE` dialplan function, or configuring various
+ features in `confbridge.conf` and `queues.conf`.
+
+
+#### uri.c: Simplify ast_uri_make_host_with_port()
+ Author: Sean Bright
+ Date: 2023-11-09
+
+
+#### func_curl.c: Remove CURLOPT() plaintext documentation.
+ Author: Sean Bright
+ Date: 2023-11-13
+
+ I assume this was missed when initially converting to XML
+ documentation and we've been kicking the can down the road since.
+
+
+#### res_http_websocket.c: Set hostname on client for certificate validation.
+ Author: Sean Bright
+ Date: 2023-11-09
+
+ Additionally add a `assert()` to in the TLS client setup code to
+ ensure that hostname is set when it is supposed to be.
+
+ Fixes #433
+
+
+#### live_ast: Add astcachedir to generated asterisk.conf.
+ Author: Sean Bright
+ Date: 2023-11-09
+
+ `astcachedir` (added in b0842713) was not added to `live_ast` so
+ continued to point to the system `/var/cache` directory instead of the
+ one in the live environment.
+
+
+#### SECURITY.md: Update with correct documentation URL
+ Author: George Joseph
+ Date: 2023-11-09
+
+
+#### func_lock: Add missing see-also refs to documentation.
+ Author: Naveen Albert
+ Date: 2023-11-09
+
+ Resolves: #423
+
+#### app_followme.c: Grab reference on nativeformats before using it
+ Author: Matthew Fredrickson
+ Date: 2023-10-25
+
+ Fixes a crash due to a lack of proper reference on the nativeformats
+ object before passing it into ast_request(). Also found potentially
+ similar use case bugs in app_chanisavail.c, bridge.c, and bridge_basic.c
+
+ Fixes: #388
+
+#### configs: Improve documentation for bandwidth in iax.conf.
+ Author: Naveen Albert
+ Date: 2023-11-09
+
+ This improves the documentation for the bandwidth setting
+ in iax.conf by making it clearer what the ramifications
+ of this setting are. It also changes the sample default
+ from low to high, since only high is compatible with good
+ codecs that people will want to use in the vast majority
+ of cases, and this is a common gotcha that trips up new users.
+
+ Resolves: #425
+
+#### logger: Add channel-based filtering.
+ Author: Naveen Albert
+ Date: 2023-08-09
+
+ This adds the ability to filter console
+ logging by channel or groups of channels.
+ This can be useful on busy systems where
+ an administrator would like to analyze certain
+ calls in detail. A dialplan function is also
+ included for the purpose of assigning a channel
+ to a group (e.g. by tenant, or some other metric).
+
+ ASTERISK-30483 #close
+
+ Resolves: #242
+
+ UserNote: The console log can now be filtered by
+ channels or groups of channels, using the
+ logger filter CLI commands.
+
+
+#### chan_iax2.c: Don't send unsanitized data to the logger.
+ Author: Sean Bright
+ Date: 2023-11-08
+
+ This resolves an issue where non-printable characters could be sent to
+ the console/log files.
+
+
+#### codec_ilbc: Disable system ilbc if version >= 3.0.0
+ Author: George Joseph
+ Date: 2023-11-07
+
+ Fedora 37 started shipping ilbc 3.0.4 which we don't yet support.
+ configure.ac now checks the system for "libilbc < 3" instead of
+ just "libilbc". If true, the system version of ilbc will be used.
+ If not, the version included at codecs/ilbc will be used.
+
+ Resolves: #84
+
+#### resource_channels.c: Explicit codec request when creating UnicastRTP.
+ Author: Sean Bright
+ Date: 2023-11-06
+
+ Fixes #394
+
+
+#### doc: Update IP Quality of Service links.
+ Author: Sean Bright
+ Date: 2023-11-07
+
+ Fixes #328
+
+
+#### chan_pjsip: Add PJSIPHangup dialplan app and manager action
+ Author: George Joseph
+ Date: 2023-10-31
+
+ See UserNote below.
+
+ Exposed the existing Hangup AMI action in manager.c so we can use
+ all of it's channel search and AMI protocol handling without
+ duplicating that code in dialplan_functions.c.
+
+ Added a lookup function to res_pjsip.c that takes in the
+ string represenation of the pjsip_status_code enum and returns
+ the actual status code. I.E. ast_sip_str2rc("DECLINE") returns
+ 603. This allows the caller to specify PJSIPHangup(decline) in
+ the dialplan, just like Hangup(call_rejected).
+
+ Also extracted the XML documentation to its own file since it was
+ almost as large as the code itself.
+
+ UserNote: A new dialplan app PJSIPHangup and AMI action allows you
+ to hang up an unanswered incoming PJSIP call with a specific SIP
+ response code in the 400 -> 699 range.
+
+
+#### chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
+ Author: Sean Bright
+ Date: 2023-11-06
+
+ When IAX2 debugging was enabled (`iax2 set debug on`), if the last IE
+ in a frame was one that may not have any data - such as the CALLTOKEN
+ IE in an NEW request - it was not getting displayed.
+
+
+#### chan_dahdi: Warn if nonexistent cadence is requested.
+ Author: Naveen Albert
+ Date: 2023-11-02
+
+ If attempting to ring a channel using a nonexistent cadence,
+ emit a warning, before falling back to the default cadence.
+
+ Resolves: #409
+
+#### stasis: Update the snapshot after setting the redirect
+ Author: Holger Hans Peter Freyther
+ Date: 2023-10-21
+
+ The previous commit added the caller_rdnis attribute. Make it
+ avialble during a possible ChanngelHangupRequest.
+
+
+#### ari: Provide the caller ID RDNIS for the channels
+ Author: Holger Hans Peter Freyther
+ Date: 2023-10-14
+
+ Provide the caller ID RDNIS when available. This will allow an
+ application to follow the redirect.
+
+
+#### main/utils: Implement ast_get_tid() for OpenBSD
+ Author: Brad Smith
+ Date: 2023-11-01
+
+ Implement the ast_get_tid() function for OpenBSD. OpenBSD supports
+ getting the TID via getthrid().
+
+
+#### res_rtp_asterisk.c: Fix runtime issue with LibreSSL
+ Author: Brad Smith
+ Date: 2023-11-02
+
+ The module will fail to load. Use proper function DTLS_method() with LibreSSL.
+
+
+#### app_directory: Add ADSI support to Directory.
+ Author: Naveen Albert
+ Date: 2023-09-27
+
+ This adds optional ADSI support to the Directory
+ application, which allows callers with ADSI CPE
+ to navigate the Directory system significantly
+ faster than is possible using the audio prompts.
+ Callers can see the directory name (and optionally
+ extension) on their screenphone and confirm or
+ reject a match immediately rather than waiting
+ for it to be spelled out, enhancing usability.
+
+ Resolves: #356
+
+#### core_local: Fix local channel parsing with slashes.
+ Author: Naveen Albert
+ Date: 2023-08-09
+
+ Currently, trying to call a Local channel with a slash
+ in the extension will fail due to the parsing of characters
+ after such a slash as being dial modifiers. Additionally,
+ core_local is inconsistent and incomplete with
+ its parsing of Local dial strings in that sometimes it
+ uses the first slash and at other times it uses the last.
+
+ For instance, something like DAHDI/5 or PJSIP/device
+ is a perfectly usable extension in the dialplan, but Local
+ channels in particular prevent these from being called.
+
+ This creates inconsistent behavior for users, since using
+ a slash in an extension is perfectly acceptable, and using
+ a Goto to accomplish this works fine, but if specified
+ through a Local channel, the parsing prevents this.
+
+ This fixes this by explicitly parsing options from the
+ last slash in the extension, rather than the first one,
+ which doesn't cause an issue for extensions with slashes.
+
+ ASTERISK-30013 #close
+
+ Resolves: #248
+
+#### Remove files that are no longer updated
+ Author: Mark Murawski
+ Date: 2023-10-30
+
+ Fixes: #360
+
+#### app_voicemail: Add AMI event for mailbox PIN changes.
+ Author: Naveen Albert
+ Date: 2023-10-30
+
+ This adds an AMI event that is emitted whenever a
+ mailbox password is successfully changed, allowing
+ AMI consumers to process these.
+
+ UserNote: The VoicemailPasswordChange event is
+ now emitted whenever a mailbox password is updated,
+ containing the mailbox information and the new
+ password.
+
+ Resolves: #398
+
+#### app_queue.c: Emit unpause reason with PauseQueueMember event.
+ Author: Sean Bright
+ Date: 2023-10-30
+
+ Fixes #395
+
+
+#### bridge_simple: Suppress unchanged topology change requests
+ Author: George Joseph
+ Date: 2023-10-30
+
+ In simple_bridge_join, we were sending topology change requests
+ even when the new and old topologies were the same. In some
+ circumstances, this can cause unnecessary re-invites and even
+ a re-invite flood. We now suppress those.
+
+ Resolves: #384
+
+#### res_pjsip: Include cipher limit in config error message.
+ Author: Naveen Albert
+ Date: 2023-10-30
+
+ If too many ciphers are specified in the PJSIP config,
+ include the maximum number of ciphers that may be
+ specified in the user-facing error message.
+
+ Resolves: #396
+
+#### res_speech: allow speech to translate input channel
+ Author: Mike Bradeen
+ Date: 2023-09-07
+
+ * Allow res_speech to translate the input channel if the
+ format is translatable to a format suppored by the
+ speech provider.
+
+ Resolves: #129
+
+ UserNote: res_speech now supports translation of an input channel
+ to a format supported by the speech provider, provided a translation
+ path is available between the source format and provider capabilites.
+
+
+#### res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
+ Author: Sean Bright
+ Date: 2023-10-25
+
+ Fixes #386
+
+
+#### res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
+ Author: Sean Bright
+ Date: 2023-10-17
+
+ Fixes #376
+
+
+#### api.wiki.mustache: Fix indentation in generated markdown
+ Author: George Joseph
+ Date: 2023-10-25
+
+ The '*' list indicator for default values and allowable values for
+ path, query and POST parameters need to be indented 4 spaces
+ instead of 2.
+
+ Should resolve issue 38 in the documentation repo.
+
+
+#### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
+ Author: Sean Bright
+ Date: 2023-10-23
+
+ Per RFC8827:
+
+ Implementations MUST NOT implement DTLS renegotiation and MUST
+ reject it with a "no_renegotiation" alert if offered.
+
+ So we disable it when webrtc=yes is set.
+
+ Fixes #378
+
+ UpgradeNote: The dtls_rekey will be disabled if webrtc support is
+ requested on an endpoint. A warning will also be emitted.
+
+
+#### configs: Fix typo in pjsip.conf.sample.
+ Author: Samuel Olaechea
+ Date: 2023-10-12
+
+
+#### res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
+ Author: George Joseph
+ Date: 2023-10-19
+
+ Commit f66f77f last year prevents the res_pjsip_exten_state and
+ res_pjsip_mwi modules from unloading due to possible pjproject
+ asserts if the modules are reloaded. A side effect of the
+ implementation is that the taskprocessors these modules use aren't
+ being released. When asterisk is doing a graceful shutdown, it
+ waits AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT seconds for all
+ taskprocessors to stop but since those 2 modules don't release
+ theirs, the shutdown hangs for that amount of time.
+
+ This change allows the modules to be unloaded and their resources to
+ be released when ast_shutdown_final is true.
+
+ Resolves: #379
+
+#### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 cha..
+ Author: sungtae kim
+ Date: 2023-09-23
+
+ This commit introduces an extension to the endpoint and relevant
+ resource sizes for PJSIP, transitioning from its current 40-character
+ constraint to a more versatile 255-character capacity. This enhancement
+ significantly overcomes limitations related to domain qualification and
+ practical usage, ultimately delivering improved functionality. In
+ addition, it includes adjustments to accommodate the expanded realm size
+ within the ARI, specifically enhancing the maximum realm length.
+
+ Resolves: #345
+
+ UserNote: With this update, the PJSIP realm lengths have been extended
+ to support up to 255 characters.
+
+ UpgradeNote: As part of this update, the maximum allowable length
+ for PJSIP endpoints and relevant resources has been increased from
+ 40 to 255 characters. To take advantage of this enhancement, it is
+ recommended to run the necessary procedures (e.g., Alembic) to
+ update your schemas.
+
+
+#### res_stasis: signal when new command is queued
+ Author: Mike Bradeen
+ Date: 2023-10-02
+
+ res_statsis's app loop sleeps for up to .2s waiting on input
+ to a channel before re-checking the command queue. This can
+ cause delays between channel setup and bridge.
+
+ This change is to send a SIGURG on the sleeping thread when
+ a new command is enqueued. This exits the sleeping thread out
+ of the ast_waitfor() call triggering the new command being
+ processed on the channel immediately.
+
+ Resolves: #362
+
+ UserNote: Call setup times should be significantly improved
+ when using ARI.
+
+
+#### ari/stasis: Indicate progress before playback on a bridge
+ Author: Holger Hans Peter Freyther
+ Date: 2023-10-02
+
+ Make it possible to start a playback and the calling party
+ to receive audio on a bridge before the call is connected.
+
+ Model the implementation after play_on_channel and deliver a
+ AST_CONTROL_PROGRESS before starting the playback.
+
+ For a PJSIP channel this will result in sending a SIP 183
+ Session Progress.
+
+
+#### func_curl.c: Ensure channel is locked when manipulating datastores.
+ Author: Sean Bright
+ Date: 2023-10-09
+
+
+#### Update config.yml
+ Author: Joshua C. Colp
+ Date: 2023-06-15
+
+
+#### logger.h: Add ability to change the prefix on SCOPE_TRACE output
+ Author: George Joseph
+ Date: 2023-10-05
+
+ You can now define the _TRACE_PREFIX_ macro to change the
+ default trace line prefix of "file:line function" to
+ something else. Full documentation in logger.h.
+
+
+#### Add libjwt to third-party
+ Author: George Joseph
+ Date: 2023-09-21
+
+ The current STIR/SHAKEN implementation is not currently usable due
+ to encryption issues. Rather than trying to futz with OpenSSL and
+ the the current code, we can take advantage of the existing
+ capabilities of libjwt but we first need to add it to the
+ third-party infrastructure already in place for jansson and
+ pjproject.
+
+ A few tweaks were also made to the third-party infrastructure as
+ a whole. The jansson "dest" install directory was renamed "dist"
+ to better match convention, and the third-party Makefile was updated
+ to clean all product directories not just the ones currently in
+ use.
+
+ Resolves: #349
+
+#### res_pjsip: update qualify_timeout documentation with DNS note
+ Author: Mike Bradeen
+ Date: 2023-09-26
+
+ The documentation on qualify_timeout does not explicitly state that the timeout
+ includes any time required to perform any needed DNS queries on the endpoint.
+
+ If the OPTIONS response is delayed due to the DNS query, it can still render an
+ endpoint as Unreachable if the net time is enough for qualify_timeout to expire.
+
+ Resolves: #352
+
+#### chan_dahdi: Clarify scope of callgroup/pickupgroup.
+ Author: Naveen Albert
+ Date: 2023-09-04
+
+ Internally, chan_dahdi only applies callgroup and
+ pickupgroup to FXO signalled channels, but this is
+ not documented anywhere. This is now documented in
+ the sample config, and a warning is emitted if a
+ user tries configuring these settings for channel
+ types that do not support these settings, since they
+ will not have any effect.
+
+ Resolves: #294
+
+#### func_json: Fix crashes for some types
+ Author: Bastian Triller
+ Date: 2023-09-21
+
+ This commit fixes crashes in JSON_DECODE() for types null, true, false
+ and real numbers.
+
+ In addition it ensures that a path is not deeper than 32 levels.
+
+ Also allow root object to be an array.
+
+ Add unit tests for above cases.
+
+
+#### res_speech_aeap: add aeap error handling
+ Author: Mike Bradeen
+ Date: 2023-09-21
+
+ res_speech_aeap previously did not register an error handler
+ with aeap, so it was not notified of a disconnect. This resulted
+ in SpeechBackground never exiting upon a websocket disconnect.
+
+ Resolves: #303
+
+#### app_voicemail: Disable ADSI if unavailable.
+ Author: Naveen Albert
+ Date: 2023-09-27
+
+ If ADSI is available on a channel, app_voicemail will repeatedly
+ try to use ADSI, even if there is no CPE that supports it. This
+ leads to many unnecessary delays during the session. If ADSI is
+ available but ADSI setup fails, we now disable it to prevent
+ further attempts to use ADSI during the session.
+
+ Resolves: #354
+
+#### codec_builtin: Use multiples of 20 for maximum_ms
+ Author: Eduardo
+ Date: 2023-07-28
+
+ Some providers require a multiple of 20 for the maxptime or fail to complete calls,
+ e.g. Vivo in Brazil. To increase compatibility, only multiples of 20 are now used.
+
+ Resolves: #260
+
+#### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
+ Author: George Joseph
+ Date: 2023-09-13
+
+ Previously, DETECT_DEADLOCKS depended on DEBUG_THREADS.
+ Unfortunately, DEBUG_THREADS adds a lot of lock tracking overhead
+ to all of the lock lifecycle calls whereas DETECT_DEADLOCKS just
+ causes the lock calls to loop over trylock in 200us intervals until
+ the lock is obtained and spits out log messages if it takes more
+ than 5 seconds. From a code perspective, the only reason they were
+ tied together was for logging. So... The ifdefs in lock.c were
+ refactored to allow DETECT_DEADLOCKS to be enabled without
+ also enabling DEBUG_THREADS.
+
+ Resolves: #321
+
+ UserNote: You no longer need to select DEBUG_THREADS to use
+ DETECT_DEADLOCKS. This removes a significant amount of overhead
+ if you just want to detect possible deadlocks vs needing full
+ lock tracing.
+
+
+#### asterisk.c: Use the euid's home directory to read/write cli history
+ Author: George Joseph
+ Date: 2023-09-15
+
+ The CLI .asterisk_history file is read from/written to the directory
+ specified by the HOME environment variable. If the root user starts
+ asterisk with the -U/-G options, or with runuser/rungroup set in
+ asterisk.conf, the asterisk process is started as root but then it
+ calls setuid/setgid to set the new user/group. This does NOT reset
+ the HOME environment variable to the new user's home directory
+ though so it's still left as "/root". In this case, the new user
+ will almost certainly NOT have access to read from or write to the
+ history file.
+
+ * Added function process_histfile() which calls
+ getpwuid(geteuid()) and uses pw->dir as the home directory
+ instead of the HOME environment variable.
+ * ast_el_read_default_histfile() and ast_el_write_default_histfile()
+ have been modified to use the new process_histfile()
+ function.
+
+ Resolves: #337
+
+#### res_pjsip_transport_websocket: Prevent transport from being destroyed before m..
+ Author: Tinet-mucw
+ Date: 2023-09-13
+
+ From the gdb information, ast_websocket_read reads a message successfully,
+ then transport_read is called in the serializer. During execution of pjsip_transport_down,
+ ws_session->stream->fd is closed; ast_websocket_read encounters an error and exits the while loop.
+ After executing transport_shutdown, the transport's reference count becomes 0, causing a crash when sending SIP messages.
+ This was due to pjsip_transport_dec_ref executing earlier than pjsip_rx_data_clone, leading to this issue.
+ In websocket_cb executeing pjsip_transport_add_ref, this we now ensure the transport is not destroyed while in the loop.
+
+ Resolves: asterisk#299
+
+#### cel: add publish user event helper
+ Author: Mike Bradeen
+ Date: 2023-09-14
+
+ Add a wrapper function around ast_cel_publish_event that
+ packs event and extras into a blob before publishing
+
+ Resolves:#330
+
+#### chan_console: Fix deadlock caused by unclean thread exit.
+ Author: Naveen Albert
+ Date: 2023-09-09
+
+ To terminate a console channel, stop_stream causes pthread_cancel
+ to make stream_monitor exit. However, commit 5b8fea93d106332bc0faa4b7fa8a6ea71e546cac
+ added locking to this function which results in deadlock due to
+ the stream_monitor thread being killed while it's holding the pvt lock.
+
+ To resolve this, a flag is now set and read to indicate abort, so
+ the use of pthread_cancel and pthread_kill can be avoided altogether.
+
+ Resolves: #308
+
+#### file.c: Add ability to search custom dir for sounds
+ Author: George Joseph
+ Date: 2023-09-11
+
+ To better co-exist with sounds files that may be managed by
+ packages, custom sound files may now be placed in
+ AST_DATA_DIR/sounds/custom instead of the standard
+ AST_DATA_DIR/sounds/<lang> directory. If the new
+ "sounds_search_custom_dir" option in asterisk.conf is set
+ to "true", asterisk will search the custom directory for sounds
+ files before searching the standard directory. For performance
+ reasons, the "sounds_search_custom_dir" defaults to "false".
+
+ Resolves: #315
+
+ UserNote: A new option "sounds_search_custom_dir" has been added to
+ asterisk.conf that allows asterisk to search
+ AST_DATA_DIR/sounds/custom for sounds files before searching the
+ standard AST_DATA_DIR/sounds/<lang> directory.
+
+
+#### chan_iax2: Improve authentication debugging.
+ Author: Naveen Albert
+ Date: 2023-08-30
+
+ Improves and adds some logging to make it easier
+ for users to debug authentication issues.
+
+ Resolves: #286
+
+#### res_rtp_asterisk: fix wrong counter management in ioqueue objects
+ Author: Vitezslav Novy
+ Date: 2023-09-05
+
+ In function rtp_ioqueue_thread_remove counter in ioqueue object is not decreased
+ which prevents unused ICE TURN threads from being removed.
+
+ Resolves: #301
+
+#### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
+ Author: George Joseph
+ Date: 2023-09-13
+
+ The previous behavior of make_buildopts_h was to not add the
+ non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS,
+ REF_DEBUG, etc. to the buildopts.h file because "it caused
+ ccache to invalidate files and extended compile times". They're
+ only defined by passing them on the gcc command line with '-D'
+ options. In practice, including them in the include file rarely
+ causes any impact because the only time ccache cares is if you
+ actually change an option so the hit occurrs only once after
+ you change it.
+
+ OK so why would we want to include them? Many IDEs follow the
+ include files to resolve defines and if the options aren't in an
+ include file, it can cause the IDE to mark blocks of "ifdeffed"
+ code as unused when they're really not.
+
+ So...
+
+ * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H
+ which tells make_buildopts_h to include the non-ABI-breaking
+ flags in buildopts.h as well as the ABI-breaking ones. The default
+ is disabled to preserve current behavior. As before though,
+ only the ABI-breaking flags appear in AST_BUILDOPTS and only
+ those are used to calculate AST_BUILDOPT_SUM.
+ A new AST_BUILDOPT_ALL define was created to capture all of the
+ flags.
+
+ * make_version_c was streamlined to use buildopts.h and also to
+ create asterisk_build_opts_all[] and ast_get_build_opts_all(void)
+
+ * "core show settings" now shows both AST_BUILDOPTS and
+ AST_BUILDOPTS_ALL.
+
+ UserNote: The "Build Options" entry in the "core show settings"
+ CLI command has been renamed to "ABI related Build Options" and
+ a new entry named "All Build Options" has been added that shows
+ both breaking and non-breaking options.
+
+
+#### func_periodic_hook: Add hangup step to avoid timeout
+ Author: Mike Bradeen
+ Date: 2023-09-12
+
+ func_periodic_hook does not hangup after playback, relying on hangup
+ which keeps the channel alive longer than necessary.
+
+ Resolves: #325
+
+#### res_stasis_recording.c: Save recording state when unmuted.
+ Author: Sean Bright
+ Date: 2023-09-12
+
+ Fixes #322
+
+
+#### res_speech_aeap: check for null format on response
+ Author: Mike Bradeen
+ Date: 2023-09-08
+
+ * Fixed issue in res_speech_aeap when unable to provide an
+ input format to check against.
+
+
+#### func_periodic_hook: Don't truncate channel name
+ Author: George Joseph
+ Date: 2023-09-11
+
+ func_periodic_hook was truncating long channel names which
+ causes issues when you need to run other dialplan functions/apps
+ on the channel.
+
+ Resolves: #319
+
+#### safe_asterisk: Change directory permissions to 755
+ Author: George Joseph
+ Date: 2023-09-11
+
+ If the safe_asterisk script detects that the /var/lib/asterisk
+ directory doesn't exist, it now creates it with 755 permissions
+ instead of 770. safe_asterisk needing to create that directory
+ should be extremely rare though because it's normally created
+ by 'make install' which already sets the permissions to 755.
+
+ Resolves: #316
+
+#### chan_rtp: Implement RTP glue for UnicastRTP channels
+ Author: Maximilian Fridrich
+ Date: 2023-09-05
+
+ Resolves: #298
+
+ UserNote: The dial string option 'g' was added to the UnicastRTP channel
+ which enables RTP glue and therefore native RTP bridges with those
+ channels.
+
+
+#### app_queue: periodic announcement configurable start time.
+ Author: Jaco Kroon
+ Date: 2023-02-21
+
+ This newly introduced periodic-announce-startdelay makes it possible to
+ configure the initial start delay of the first periodic announcement
+ after which periodic-announce-frequency takes over.
+
+ UserNote: Introduce a new queue configuration option called
+ 'periodic-announce-startdelay' which will vary the normal (historic)
+ behavior of starting the periodic announcement cycle at
+ periodic-announce-frequency seconds after entering the queue to start
+ the periodic announcement cycle at period-announce-startdelay seconds
+ after joining the queue. The default behavior if this config option is
+ not set remains unchanged.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### variables: Add additional variable dialplan functions.
+ Author: Joshua C. Colp
+ Date: 2023-08-31
+
+ Using the Set dialplan application does not actually
+ delete channel or global variables. Instead the
+ variables are set to an empty value.
+
+ This change adds two dialplan functions,
+ GLOBAL_DELETE and DELETE which can be used to
+ delete global and channel variables instead
+ of just setting them to empty.
+
+ There is also no ability within the dialplan to
+ determine if a global or channel variable has
+ actually been set or not.
+
+ This change also adds two dialplan functions,
+ GLOBAL_EXISTS and VARIABLE_EXISTS which can be
+ used to determine if a global or channel variable
+ has been set or not.
+
+ Resolves: #289
+
+ UserNote: Four new dialplan functions have been added.
+ GLOBAL_DELETE and DELETE have been added which allows
+ the deletion of global and channel variables.
+ GLOBAL_EXISTS and VARIABLE_EXISTS have been added
+ which checks whether a global or channel variable has
+ been set.
+
+
+#### Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
+ Author: George Joseph
+ Date: 2024-01-12
+
+
+#### res_rtp_asterisk: Fix regression issues with DTLS client check
+ Author: George Joseph
+ Date: 2023-12-15
+
+ * Since ICE candidates are used for the check and pjproject is
+ required to use ICE, res_rtp_asterisk was failing to compile
+ when pjproject wasn't available. The check is now wrapped
+ with an #ifdef HAVE_PJPROJECT.
+
+ * The rtp->ice_active_remote_candidates container was being
+ used to check the address on incoming packets but that
+ container doesn't contain peer reflexive candidates discovered
+ during negotiation. This was causing the check to fail
+ where it shouldn't. We now check against pjproject's
+ real_ice->rcand array which will contain those candidates.
+
+ * Also fixed a bug in ast_sockaddr_from_pj_sockaddr() where
+ we weren't zeroing out sin->sin_zero before returning. This
+ was causing ast_sockaddr_cmp() to always return false when
+ one of the inputs was converted from a pj_sockaddr, even
+ if both inputs had the same address and port.
+
+ Resolves: #500
+ Resolves: #503
+ Resolves: #505
+
+#### res_pjsip_header_funcs: Duplicate new header value, don't copy.
+ Author: Gitea
+ Date: 2023-07-10
+
+ When updating an existing header the 'update' code incorrectly
+ just copied the new value into the existing buffer. If the
+ new value exceeded the available buffer size memory outside
+ of the buffer would be written into, potentially causing
+ a crash.
+
+ This change makes it so that the 'update' now duplicates
+ the new header value instead of copying it into the existing
+ buffer.
+
+#### res_pjsip: disable raw bad packet logging
+ Author: Mike Bradeen
+ Date: 2023-07-25
+
+ Add patch to split the log level for invalid packets received on the
+ signaling port. The warning regarding the packet will move to level 2
+ so that it can still be displayed, while the raw packet will be at level
+ 4.
+
+#### res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
+ Author: George Joseph
+ Date: 2023-11-09
+
+ When ICE is in use, we can prevent a possible DOS attack by allowing
+ DTLS protocol messages (client hello, etc) only from sources that
+ are in the active remote candidates list.
+
+ Resolves: GHSA-hxj9-xwr8-w8pq
+
+#### manager.c: Prevent path traversal with GetConfig.
+ Author: Ben Ford
+ Date: 2023-11-13
+
+ When using AMI GetConfig, it was possible to access files outside of the
+ Asterisk configuration directory by using filenames with ".." and "./"
+ even while live_dangerously was not enabled. This change resolves the
+ full path and ensures we are still in the configuration directory before
+ attempting to access the file.
+
+#### ari-stubs: Fix more local anchor references
+ Author: George Joseph
+ Date: 2023-09-05
+
+ Also allow CreateDocs job to be run manually with default branches.
+
+
+#### ari-stubs: Fix more local anchor references
+ Author: George Joseph
+ Date: 2023-09-05
+
+ Also allow CreateDocs job to be run manually with default branches.
+
+
+#### ari-stubs: Fix broken documentation anchors
+ Author: George Joseph
+ Date: 2023-09-05
+
+ All of the links that reference page anchors with capital letters in
+ the ids (#Something) have been changed to lower case to match the
+ anchors that are generated by mkdocs.
+
+
+#### res_pjsip_session: Send Session Interval too small response
+ Author: Bastian Triller
+ Date: 2023-08-28
+
+ Handle session interval lower than endpoint's configured minimum timer
+ when sending first answer. Timer setting is checked during this step and
+ needs to handled appropriately.
+ Before this change, no response was sent at all. After this change a
+ response with 422 Session Interval too small is sent to UAC.
+
+
+#### app_dial: Fix infinite loop when sending digits.
+ Author: Naveen Albert
+ Date: 2023-08-28
+
+ If the called party hangs up while digits are being
+ sent, -1 is returned to indicate so, but app_dial
+ was not checking the return value, resulting in
+ the hangup being lost and looping forever until
+ the caller manually hangs up the channel. We now
+ abort if digit sending fails.
+
+ ASTERISK-29428 #close
+
+ Resolves: #281
+
+#### app_voicemail: Fix for loop declarations
+ Author: Mike Bradeen
+ Date: 2023-08-29
+
+ Resolve for loop initial declarations added in cli changes.
+
+ Resolves: #275
+
+#### alembic: Fix quoting of the 100rel column
+ Author: George Joseph
+ Date: 2023-08-28
+
+ Add quoting around the ps_endpoints 100rel column in the ALTER
+ statements. Although alembic doesn't complain when generating
+ sql statements, postgresql does (rightly so).
+
+ Resolves: #274
+
+#### pbx.c: Fix gcc 12 compiler warning.
+ Author: Naveen Albert
+ Date: 2023-08-27
+
+ Resolves: #277
+
+#### app_audiosocket: Fixed timeout with -1 to avoid busy loop.
+ Author: zhengsh
+ Date: 2023-08-24
+
+ Resolves: asterisk#234
+
+#### download_externals: Fix a few version related issues
+ Author: George Joseph
+ Date: 2023-08-18
+
+ * Fixed issue with the script not parsing the new tag format for
+ certified releases. The format changed from certified/18.9-cert5
+ to certified-18.9-cert5.
+
+ * Fixed issue where the asterisk version wasn't being considered
+ when looking for cached versions.
+
+ Resolves: #263
+
+#### main/refer.c: Fix double free in refer_data_destructor + potential leak
+ Author: Maximilian Fridrich
+ Date: 2023-08-21
+
+ Resolves: #267
+
+#### sig_analog: Add Called Subscriber Held capability.
+ Author: Naveen Albert
+ Date: 2023-08-09
+
+ This adds support for Called Subscriber Held for FXS
+ lines, which allows users to go on hook when receiving
+ a call and resume the call later from another phone on
+ the same line, without disconnecting the call. This is
+ a convenience mechanism that most real PSTN telephone
+ switches support.
+
+ ASTERISK-30372 #close
+
+ Resolves: #240
+
+ UserNote: Called Subscriber Held is now supported for analog
+ FXS channels, using the calledsubscriberheld option. This allows
+ a station user to go on hook when receiving an incoming call
+ and resume from another phone on the same line by going on hook,
+ without disconnecting the call.
+
+
+#### app_macro: Fix locking around datastore access
+ Author: Matthew Fredrickson
+ Date: 2023-08-21
+
+ app_macro sometimes would crash due to datastore list corruption on the
+ channel because of lack of locking around find and create process for
+ the macro datastore. This patch locks the channel lock prior to protect
+ against this problem.
+
+ Resolves: #265
+
+#### Revert "app_stack: Print proper exit location for PBXless channels."
+ Author: Matthew Fredrickson
+ Date: 2023-08-10
+
+ This reverts commit 617dad4cba1513dddce87b8e95a61415fb587cf1.
+
+ apps/app_stack.c: Revert buggy gosub patch
+
+ This seems to break the case when a predial macro calls a gosub.
+ When the gosub calls return, the Return function outputs:
+
+ app_stack.c:423 return_exec: Return without Gosub: stack is empty
+
+ This returns -1 to the calling macro, which returns to app_dial
+ and causes the call to hangup instead of proceeding with the macro
+ that invoked the gosub.
+
+ Resolves: #253
+
+#### install_prereq: Fix dependency install on aarch64.
+ Author: Jason D. McCormick
+ Date: 2023-04-28
+
+ Fixes dependency solutions in install_prereq for Debian aarch64
+ platforms. install_prereq was attempting to forcibly install 32-bit
+ armhf packages due to the aptitude search for dependencies.
+
+ Resolves: #37
+
+#### res_pjsip.c: Set contact_user on incoming call local Contact header
+ Author: MikeNaso
+ Date: 2023-08-08
+
+ If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
+
+ Resolves: #226
+
+#### extconfig: Allow explicit DB result set ordering to be disabled.
+ Author: Sean Bright
+ Date: 2023-07-12
+
+ Added a new boolean configuration flag -
+ `order_multi_row_results_by_initial_column` - to both res_pgsql.conf
+ and res_config_odbc.conf that allows the administrator to disable the
+ explicit `ORDER BY` that was previously being added to all generated
+ SQL statements that returned multiple rows.
+
+ Fixes: #179
+
+#### rest-api: Run make ari-stubs
+ Author: George Joseph
+ Date: 2023-08-09
+
+ An earlier cherry-pick that involved rest-api somehow didn't include
+ a comment change in res/ari/resource_endpoints.h. This commit
+ corrects that. No changes other than the comment.
+
+
+#### res_pjsip_header_funcs: Make prefix argument optional.
+ Author: Naveen Albert
+ Date: 2023-08-09
+
+ The documentation for PJSIP_HEADERS claims that
+ prefix is optional, but in the code it is actually not.
+ However, there is no inherent reason for this, as users
+ may want to retrieve all header names, not just those
+ beginning with a certain prefix.
+
+ This makes the prefix optional for this function,
+ simply fetching all header names if not specified.
+ As a result, the documentation is now correct.
+
+ Resolves: #230
+
+ UserNote: The prefix argument to PJSIP_HEADERS is now
+ optional. If not specified, all header names will be
+ returned.
+
+
+#### pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
+ Author: George Joseph
+ Date: 2023-08-11
+
+ The default is 32 with 8 being used by pjproject itself. Recent
+ commits have put us over the limit resulting in assertions in
+ pjproject. Since this value is used in invites, dialogs,
+ transports and subscriptions as well as the global pjproject
+ endpoint, we don't want to increase it too much.
+
+ Resolves: #255
+
+#### manager: Tolerate stasis messages with no channel snapshot.
+ Author: Joshua C. Colp
+ Date: 2023-08-09
+
+ In some cases I have yet to determine some stasis messages may
+ be created without a channel snapshot. This change adds some
+ tolerance to this scenario, preventing a crash from occurring.
+
+
+#### core/ari/pjsip: Add refer mechanism
+ Author: Maximilian Fridrich
+ Date: 2023-05-10
+
+ This change adds support for refers that are not session based. It
+ includes a refer implementation for the PJSIP technology which results
+ in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
+ triggered using the new ARI endpoint `/endpoints/refer`.
+
+ Resolves: #71
+
+ UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
+ an endpoint to some URI or endpoint.
+
+
+#### chan_dahdi: Allow autoreoriginating after hangup.
+ Author: Naveen Albert
+ Date: 2023-08-04
+
+ Currently, if an FXS channel is still off hook when
+ all calls on the line have hung up, the user is provided
+ reorder tone until going back on hook again.
+
+ In addition to not reflecting what most commercial switches
+ actually do, it's very common for switches to automatically
+ reoriginate for the user so that dial tone is provided without
+ the user having to depress and release the hookswitch manually.
+ This can increase convenience for users.
+
+ This behavior is now supported for kewlstart FXS channels.
+ It's supported only for kewlstart (FXOKS) mainly because the
+ behavior doesn't make any sense for ground start channels,
+ and loop start signalling doesn't provide the necessary DAHDI
+ event that makes this easy to implement. Likely almost everyone
+ is using FXOKS over FXOLS anyways since FXOLS is pretty useless
+ these days.
+
+ ASTERISK-30357 #close
+
+ Resolves: #224
+
+ UserNote: The autoreoriginate setting now allows for kewlstart FXS
+ channels to automatically reoriginate and provide dial tone to the
+ user again after all calls on the line have cleared. This saves users
+ from having to manually hang up and pick up the receiver again before
+ making another call.
+
+
+#### audiohook: Unlock channel in mute if no audiohooks present.
+ Author: Joshua C. Colp
+ Date: 2023-08-09
+
+ In the case where mute was called on a channel that had no
+ audiohooks the code was not unlocking the channel, resulting
+ in a deadlock.
+
+ Resolves: #233
+
+#### sig_analog: Allow three-way flash to time out to silence.
+ Author: Naveen Albert
+ Date: 2023-07-10
+
+ sig_analog allows users to flash and use the three-way dial
+ tone as a primitive hold function, simply by never timing
+ it out.
+
+ Some systems allow this dial tone to time out to silence,
+ so the user is not annoyed by a persistent dial tone.
+ This option allows the dial tone to time out normally to
+ silence.
+
+ ASTERISK-30004 #close
+ Resolves: #205
+
+ UserNote: The threewaysilenthold option now allows the three-way
+ dial tone to time out to silence, rather than continuing forever.
+
+
+#### res_prometheus: Do not generate broken metrics
+ Author: Holger Hans Peter Freyther
+ Date: 2023-04-07
+
+ In 8d6fdf9c3adede201f0ef026dab201b3a37b26b6 invisible bridges were
+ skipped but that lead to producing metrics with no name and no help.
+
+ Keep track of the number of metrics configured and then only emit these.
+ Add a basic testcase that verifies that there is no '(NULL)' in the
+ output.
+
+ ASTERISK-30474
+
+
+#### res_pjsip: Enable TLS v1.3 if present.
+ Author: Sean Bright
+ Date: 2023-08-02
+
+ Fixes #221
+
+ UserNote: res_pjsip now allows TLS v1.3 to be enabled if supported by
+ the underlying PJSIP library. The bundled version of PJSIP supports
+ TLS v1.3.
+
+
+#### func_cut: Add example to documentation.
+ Author: phoneben
+ Date: 2023-07-19
+
+ This adds an example to the XML documentation clarifying usage
+ of the CUT function to address a common misusage.
+
+
+#### extensions.conf.sample: Remove reference to missing context.
+ Author: Sean Bright
+ Date: 2023-07-16
+
+ c3ff4648 removed the [iaxtel700] context but neglected to remove
+ references to it.
+
+ This commit addresses that and also removes iaxtel and freeworlddialup
+ references from other config files.
+
+
+#### func_export: Use correct function argument as variable name.
+ Author: Sean Bright
+ Date: 2023-07-12
+
+ Fixes #208
+
+
+#### app_queue: Add support for applying caller priority change immediately.
+ Author: Joshua C. Colp
+ Date: 2023-07-07
+
+ The app_queue module provides both an AMI action and a CLI command
+ to change the priority of a caller in a queue. Up to now this change
+ of priority has only been reflected to new callers into the queue.
+
+ This change adds an "immediate" option to both the AMI action and
+ CLI command which immediately applies the priority change respective
+ to the other callers already in the queue. This can allow, for example,
+ a caller to be placed at the head of the queue immediately if their
+ priority is sufficient.
+
+ Resolves: #202
+
+ UserNote: The 'queue priority caller' CLI command and
+ 'QueueChangePriorityCaller' AMI action now have an 'immediate'
+ argument which allows the caller priority change to be reflected
+ immediately, causing the position of a caller to move within the
+ queue depending on the priorities of the other callers.
+
+
+#### chan_iax2.c: Avoid crash with IAX2 switch support.
+ Author: Sean Bright
+ Date: 2023-07-07
+
+ A change made in 82cebaa0 did not properly handle the case when a
+ channel was not provided, triggering a crash. ast_check_hangup(...)
+ does not protect against NULL pointers.
+
+ Fixes #180
+
+
+#### res_geolocation: Ensure required 'location_info' is present.
+ Author: Sean Bright
+ Date: 2023-07-07
+
+ Fixes #189
+
+
+#### Adds manager actions to allow move/remove/forward individual messages in a par..
+ Author: Mike Bradeen
+ Date: 2023-06-29
+
+ Resolves: #181
+
+ UserNote: The following manager actions have been added
+
+ VoicemailBoxSummary - Generate message list for a given mailbox
+
+ VoicemailRemove - Remove a message from a mailbox folder
+
+ VoicemailMove - Move a message from one folder to another within a mailbox
+
+ VoicemailForward - Copy a message from one folder in one mailbox
+ to another folder in another or the same mailbox.
+
+
+#### app_voicemail: add CLI commands for message manipulation
+ Author: Mike Bradeen
+ Date: 2023-06-20
+
+ Adds CLI commands to allow move/remove/forward individual messages
+ from a particular mailbox folder. The forward command can be used
+ to copy a message within a mailbox or to another mailbox. Also adds
+ a show mailbox, required to retrieve message ID's.
+
+ Resolves: #170
+
+ UserNote: The following CLI commands have been added to app_voicemail
+
+ voicemail show mailbox <mailbox> <context>
+ Show contents of mailbox <mailbox>@<context>
+
+ voicemail remove <mailbox> <context> <from_folder> <messageid>
+ Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
+
+ voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
+ Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
+
+ voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
+ Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
+ mailbox <mailbox>@<context> <to_folder>
+
+
+#### res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` i..
+ Author: zhengsh
+ Date: 2023-06-30
+
+ From the gdb information, it was found that when calling __ast_free, the size of the
+ allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
+ is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
+ it is found to be 1.
+
+ Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
+ which is outside the protection of the rtp_instance lock. However,
+ ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
+ rtp->themssrc_valid within the protection of the rtp_instance lock.
+
+ This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
+ ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
+ within ast_rtcp_generate_report().
+
+ Resolves: asterisk#63
+
+#### sig_analog: Allow immediate fake ring to be suppressed.
+ Author: Naveen Albert
+ Date: 2023-06-08
+
+ When immediate=yes on an FXS channel, sig_analog will
+ start fake audible ringback that continues until the
+ channel is answered. Even if it answers immediately,
+ the ringback is still audible for a brief moment.
+ This can be disruptive and unwanted behavior.
+
+ This adds an option to disable this behavior, though
+ the default behavior remains unchanged.
+
+ ASTERISK-30003 #close
+ Resolves: #118
+
+ UserNote: The immediatering option can now be set to no to suppress
+ the fake audible ringback provided when immediate=yes on FXS channels.
+
+
+#### app.h: Move declaration of ast_getdata_result before its first use
+ Author: George Joseph
+ Date: 2023-07-10
+
+ The ast_app_getdata() and ast_app_getdata_terminator() declarations
+ in app.h were changed recently to return enum ast_getdata_result
+ (which is how they were defined in app.c). The existing
+ declaration of ast_getdata_result in app.h was about 1000 lines
+ after those functions however so under certain circumstances,
+ a "use before declaration" error was thrown by the compiler.
+ The declaration of the enum was therefore moved to before those
+ functions.
+
+ Resolves: #200
+
+#### doc: Remove obsolete CHANGES-staging and UPGRADE-staging
+ Author: George Joseph
+ Date: 2023-07-10
+
+
+#### app_voicemail: fix imap compilation errors
+ Author: Mike Bradeen
+ Date: 2023-06-26
+
+ Fixes two compilation errors in app_voicemail_imap, one due to an unsed
+ variable and one due to a new variable added in the incorrect location
+ in _163.
+
+ Resolves: #174
+
+#### res_musiconhold: avoid moh state access on unlocked chan
+ Author: Mike Bradeen
+ Date: 2023-05-31
+
+ Move channel unlock to after moh state access to avoid
+ potential unlocked access to state.
+
+ Resolves: #133
+
+#### utils: add lock timestamps for DEBUG_THREADS
+ Author: Mike Bradeen
+ Date: 2023-05-23
+
+ Adds last locked and unlocked timestamps as well as a
+ counter for the number of times the lock has been
+ attempted (vs locked/unlocked) to debug output printed
+ using the DEBUG_THREADS option.
+
+ Resolves: #110
+
+#### rest-api: Updates for new documentation site
+ Author: George Joseph
+ Date: 2023-06-26
+
+ The new documentation site uses traditional markdown instead
+ of the Confluence flavored version. This required changes in
+ the mustache templates and the python that generates the files.
+
+
+#### app_voicemail_imap: Fix message count when IMAP server is unavailable
+ Author: Olaf Titz
+ Date: 2023-06-15
+
+ Some callers of __messagecount did not correctly handle error return,
+ instead returning a -1 message count.
+ This caused a notification with "Messages-Waiting: yes" and
+ "Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable.
+
+ Fixes: #64
+
+#### res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
+ Author: Sean Bright
+ Date: 2023-06-12
+
+ Resolves: #116
+
+#### res_pjsip_session: Added new function calls to avoid ABI issues.
+ Author: Ben Ford
+ Date: 2023-06-05
+
+ Added two new functions (ast_sip_session_get_dialog and
+ ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the
+ pjsip_inv_state respectively from the pjsip_inv_session on the
+ ast_sip_session struct. This is due to pjproject adding a new field to
+ the pjsip_inv_session struct that caused crashes when trying to access
+ fields that were no longer where they were expected to be if a module
+ was compiled against a different version of pjproject.
+
+ Resolves: #145
+
+#### app_queue: Add force_longest_waiting_caller option.
+ Author: Nathan Bruning
+ Date: 2023-01-24
+
+ This adds an option 'force_longest_waiting_caller' which changes the
+ global behavior of the queue engine to prevent queue callers from
+ 'jumping ahead' when an agent is in multiple queues.
+
+ Resolves: #108
+
+ Also closes old asterisk issues:
+ - ASTERISK-17732
+ - ASTERISK-17570
+
+
+#### pjsip_transport_events.c: Use %zu printf specifier for size_t.
+ Author: Sean Bright
+ Date: 2023-06-05
+
+ Partially resolves #143.
+
+
+#### res_crypto.c: Gracefully handle potential key filename truncation.
+ Author: Sean Bright
+ Date: 2023-06-05
+
+ Partially resolves #143.
+
+
+#### configure: Remove obsolete and deprecated constructs.
+ Author: Sean Bright
+ Date: 2023-06-01
+
+ These were uncovered when trying to run `bootstrap.sh` with Autoconf
+ 2.71:
+
+ * AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS().
+ * AC_HEADER_TIME is obsolete.
+ * $as_echo is deprecated in favor of AS_ECHO() which requires an update
+ to ax_pthread.m4.
+
+ Note that the generated artifacts in this commit are from Autoconf 2.69.
+
+ Resolves #139
+
+
+#### res_fax_spandsp.c: Clean up a spaces/tabs issue
+ Author: zhou_jiajian
+ Date: 2023-05-26
+
+
+#### ast-db-manage: Synchronize revisions between comments and code.
+ Author: Sean Bright
+ Date: 2023-06-06
+
+ In a handful of migrations, the comment header that indicates the
+ current and previous revisions has drifted from the identifiers
+ revision and down_revision variables. This updates the comment headers
+ to match the code.
+
+
+#### test_statis_endpoints: Fix channel_messages test again
+ Author: George Joseph
+ Date: 2023-06-12
+
+
+#### res_crypto.c: Avoid using the non-portable ALLPERMS macro.
+ Author: Sean Bright
+ Date: 2023-06-05
+
+ ALLPERMS is not POSIX and it's trivial enough to not jump through
+ autoconf hoops to check for it.
+
+ Fixes #149.
+
+
+#### tcptls: when disabling a server port, we should set the accept_fd to -1.
+ Author: Jaco Kroon
+ Date: 2023-06-02
+
+ If we don't set this to -1 if the structure can be potentially re-used
+ later then it's possible that we'll issue a close() on an unrelated file
+ descriptor, breaking asterisk in other interesting ways.
+
+ I believe this to be an unlikely scenario, but it costs nothing to be
+ safe.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### AMI: Add parking position parameter to Park action
+ Author: Jiajian Zhou
+ Date: 2023-05-19
+
+ Add a parking space extension parameter (ParkingSpace) to the Park action.
+ Park action will attempt to park the call to that extension.
+ If the extension is already in use, then execution will continue at the next priority.
+
+ UserNote: New ParkingSpace parameter has been added to AMI action Park.
+
+#### test_stasis_endpoints.c: Make channel_messages more stable
+ Author: George Joseph
+ Date: 2023-06-09
+
+ The channel_messages test was assuming that stasis would return
+ messages in a specific order. This is an incorrect assumption as
+ message ordering was never guaranteed. This was causing the test
+ to fail occasionally. We now test all the messages for the
+ required message types instead of testing one by one.
+
+ Resolves: #158
+
+#### build: Fix a few gcc 13 issues
+ Author: George Joseph
+ Date: 2023-06-09
+
+ * gcc 13 is now catching when a function is declared as returning
+ an enum but defined as returning an int or vice versa. Fixed
+ a few in app.h, loader.c, stasis_message.c.
+
+ * gcc 13 is also now (incorrectly) complaining of dangling pointers
+ when assigning a pointer to a local char array to a char *. Had
+ to change that to an ast_alloca.
+
+ Resolves: #155
+
+#### ast-db-manage: Fix alembic branching error caused by #122.
+ Author: Sean Bright
+ Date: 2023-06-05
+
+ Fixes #147.
+
+
+#### app_followme: fix issue with enable_callee_prompt=no (#88)
+ Author: alex2grad
+ Date: 2023-06-05
+
+ * app_followme: fix issue with enable_callee_prompt=no
+
+ If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk
+ incorrectly sets a winner channel to the channel from which any control frame was read.
+
+ This fix sets the winner channel only to the answered channel.
+
+ Resolves: #87
+
+ ASTERISK-30326
+
+
+#### sounds: Update download URL to use HTTPS.
+ Author: Sean Bright
+ Date: 2023-06-01
+
+ Related to #136
+
+
+#### configure: Makefile downloader enable follow redirects.
+ Author: Miguel Angel Nubla
+ Date: 2023-06-01
+
+ If curl is used for building, any download such as a sounds package
+ will fail to follow HTTP redirects and will download wrong data.
+
+ Resolves: #136
+
+#### res_musiconhold: Add option to loop last file.
+ Author: Naveen Albert
+ Date: 2023-05-25
+
+ Adds the loop_last option to res_musiconhold,
+ which allows the last audio file in the directory
+ to be looped perpetually once reached, rather than
+ circling back to the beginning again.
+
+ Resolves: #122
+ ASTERISK-30462
+
+ UserNote: The loop_last option in musiconhold.conf now
+ allows the last file in the directory to be looped once reached.
+
+
+#### chan_dahdi: Fix Caller ID presentation for FXO ports.
+ Author: Naveen Albert
+ Date: 2023-05-25
+
+ Currently, the presentation for incoming channels is
+ always available, because it is never actually set,
+ meaning the channel presentation can be nonsensical.
+ If the presentation from the incoming Caller ID spill
+ is private or unavailable, we now update the channel
+ presentation to reflect this.
+
+ Resolves: #120
+ ASTERISK-30333
+ ASTERISK-21741
+
+
+#### AMI: Add CoreShowChannelMap action.
+ Author: Ben Ford
+ Date: 2023-05-18
+
+ Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
+ and provides a list of all channels that are connected to that channel,
+ following local channel connections as well.
+
+ Resolves: #104
+
+ UserNote: New AMI action CoreShowChannelMap has been added.
+
+#### sig_analog: Add fuller Caller ID support.
+ Author: Naveen Albert
+ Date: 2023-05-18
+
+ A previous change, ASTERISK_29991, made it possible
+ to send additional Caller ID parameters that were
+ not previously supported.
+
+ This change adds support for analog DAHDI channels
+ to now be able to receive these parameters for
+ on-hook Caller ID, in order to enhance the usability
+ of CPE that support these parameters.
+
+ Resolves: #94
+ ASTERISK-30331
+
+ UserNote: Additional Caller ID properties are now supported on
+ incoming calls to FXS stations, namely the
+ redirecting reason and call qualifier.
+
+
+#### res_stasis.c: Add new type 'sdp_label' for bridge creation.
+ Author: Joe Searle
+ Date: 2023-05-25
+
+ Add new type 'sdp_label' when creating a bridge using the ARI. This will
+ add labels to the SDP for each stream, the label is set to the
+ corresponding channel id.
+
+ Resolves: #91
+
+ UserNote: When creating a bridge using the ARI the 'type' argument now
+ accepts a new value 'sdp_label' which will configure the bridge to add
+ labels for each stream in the SDP with the corresponding channel id.
+
+
+#### app_queue: Preserve reason for realtime queues
+ Author: Niklas Larsson
+ Date: 2023-05-05
+
+ When Asterisk is restarted it does not preserve paused reason for
+ members of realtime queues. This was fixed for non-realtime queues in
+ ASTERISK_25732
+
+ Resolves: #66
+
+ UpgradeNote: Add a new column to the queue_member table:
+ reason_paused VARCHAR(80) so the reason can be preserved.
+
+ UserNote: Make paused reason in realtime queues persist an
+ Asterisk restart. This was fixed for non-realtime
+ queues in ASTERISK_25732.
+
+
+#### indications: logging changes
+ Author: Mike Bradeen
+ Date: 2023-05-16
+
+ Increase verbosity to indicate failure due to missing country
+ and to specify default on CLI dump
+
+ Resolves: #89
+
+#### callerid: Allow specifying timezone for date/time.
+ Author: Naveen Albert
+ Date: 2023-05-18
+
+ The Caller ID generation routine currently is hardcoded
+ to always use the system time zone. This makes it possible
+ to optionally specify any TZ-format time zone.
+
+ Resolves: #98
+ ASTERISK-30330
+
+
+#### logrotate: Fix duplicate log entries.
+ Author: Naveen Albert
+ Date: 2023-05-18
+
+ The Asterisk logrotate script contains explicit
+ references to files with the .log extension,
+ which are also included when *log is expanded.
+ This causes issues with newer versions of logrotate.
+ This fixes this by ensuring that a log file cannot
+ be referenced multiple times after expansion occurs.
+
+ Resolves: #96
+ ASTERISK-30442
+ Reported by: EN Barnett
+ Tested by: EN Barnett
+
+
+#### chan_pjsip: Allow topology/session refreshes in early media state
+ Author: Maximilian Fridrich
+ Date: 2023-05-10
+
+ With this change, session modifications in the early media state are
+ possible if the SDP was sent reliably and confirmed by a PRACK. For
+ details, see RFC 6337, escpecially section 3.2.
+
+ Resolves: #73
+
+#### chan_dahdi: Fix broken hidecallerid setting.
+ Author: Naveen Albert
+ Date: 2023-05-18
+
+ The hidecallerid setting in chan_dahdi.conf currently
+ is broken for a couple reasons.
+
+ First, the actual code in sig_analog to "allow" or "block"
+ Caller ID depending on this setting improperly used
+ ast_set_callerid instead of updating the presentation.
+ This issue was mostly fixed in ASTERISK_29991, and that
+ fix is carried forward to this code as well.
+
+ Secondly, the hidecallerid setting is set on the DAHDI
+ pvt but not carried forward to the analog pvt properly.
+ This is because the chan_dahdi config loading code improperly
+ set permhidecallerid to permhidecallerid from the config file,
+ even though hidecallerid is what is actually set from the config
+ file. (This is done correctly for call waiting, a few lines above.)
+ This is fixed to read the proper value.
+
+ Thirdly, in sig_analog, hidecallerid is set to permhidecallerid
+ only on hangup. This can lead to potential security vulnerabilities
+ as an allowed Caller ID from an initial call can "leak" into subsequent
+ calls if no hangup occurs between them. This is fixed by setting
+ hidecallerid to permcallerid when calls begin, rather than when they end.
+ This also means we don't need to also set hidecallerid in chan_dahdi.c
+ when copying from the config, as we would have to otherwise.
+
+ Fourthly, sig_analog currently only allows dialing *67 or *82 if
+ that would actually toggle the presentation. A comment is added
+ clarifying that this behavior is okay.
+
+ Finally, a couple log messages are updated to be more accurate.
+
+ Resolves: #100
+ ASTERISK-30349 #close
+
+
+#### asterisk.c: Fix option warning for remote console.
+ Author: Naveen Albert
+ Date: 2023-05-18
+
+ Commit 09e989f972e2583df4e9bf585c246c37322d8d2f
+ categorized the T option as not being compatible
+ with remote consoles, but they do affect verbose
+ messages with remote console. This fixes this.
+
+ Resolves: #102
+
+#### configure: fix test code to match gethostbyname_r prototype.
+ Author: Jaco Kroon
+ Date: 2023-05-10
+
+ This enables the test to work with CC=clang.
+
+ Without this the test for 6 args would fail with:
+
+ utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration
+ static int gethostbyname_r (const char *name, struct hostent *ret, char *buf,
+ ^
+ /usr/include/netdb.h:177:12: note: previous declaration is here
+ extern int gethostbyname_r (const char *__restrict __name,
+ ^
+
+ Fixing the expected return type to int sorts this out.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
+ Author: Sean Bright
+ Date: 2023-05-11
+
+ The functionality we are interested in is present only in pjsip 2.13
+ and newer.
+
+ Resolves: #45
+
+#### res_sorcery_memory_cache.c: Fix memory leak
+ Author: zhengsh
+ Date: 2023-05-03
+
+ Replace the original call to ast_strdup with a call to ast_strdupa to fix the leak issue.
+
+ Resolves: #55
+ ASTERISK-30429
+
+
+#### xml.c: Process XML Inclusions recursively.
+ Author: Sean Bright
+ Date: 2023-05-09
+
+ If processing an XInclude results in new <xi:include> elements, we
+ need to run XInclude processing again. This continues until no
+ replacement occurs or an error is encountered.
+
+ There is a separate issue with dynamic strings (ast_str) that will be
+ addressed separately.
+
+ Resolves: #65
+
+#### apply_patches: Use globbing instead of file/sort.
+ Author: Sean Bright
+ Date: 2023-07-06
+
+ This accomplishes the same thing as a `find ... | sort` but with the
+ added benefit of clarity and avoiding a call to a subshell.
+
+ Additionally drop the -s option from call to patch as it is not POSIX.
+
+#### apply_patches: Sort patch list before applying
+ Author: George Joseph
+ Date: 2023-07-06
+
+ The apply_patches script wasn't sorting the list of patches in
+ the "patches" directory before applying them. This left the list
+ in an indeterminate order. In most cases, the list is actually
+ sorted but rarely, they can be out of order and cause dependent
+ patches to fail to apply.
+
+ We now sort the list but the "sort" program wasn't in the
+ configure scripts so we needed to add that and regenerate
+ the scripts as well.
+
+ Resolves: #193
+
+#### pjsip: Upgrade bundled version to pjproject 2.13.1
+ Author: Stanislav Abramenkov
+ Date: 2023-07-05
+
+
+#### Set up new ChangeLogs directory
+ Author: George Joseph
+ Date: 2023-05-09
+
+
+#### chan_pjsip: also return all codecs on empty re-INVITE for late offers
+ Author: Henning Westerholt
+ Date: 2023-05-03
+
+ We should also return all codecs on an re-INVITE without SDP for a
+ call that used late offer (e.g. no SDP in the initial INVITE, SDP
+ in the ACK). Bugfix for feature introduced in ASTERISK-30193
+ (https://issues.asterisk.org/jira/browse/ASTERISK-30193)
+
+ Migration from previous gerrit change that was not merged.
+
+
+#### cel: add local optimization begin event
+ Author: Mike Bradeen
+ Date: 2023-05-02
+
+ The current AST_CEL_LOCAL_OPTIMIZE event is and has been
+ triggered on a local optimization end to serve as a flag
+ indicating the event occurred. This change adds a second
+ AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
+
+ Resolves: #52
+
+ UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
+ to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
+ can be ignored if desired.
+
+ UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
+ by itself or in conert with the existing
+ AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
+
+
+#### core: Cleanup gerrit and JIRA references. (#57)
+ Author: Sean Bright
+ Date: 2023-05-03
+
+ * Remove .gitreview and switch to pulling the main asterisk branch
+ version from configure.ac instead.
+
+ * Replace references to JIRA with GitHub.
+
+ * Other minor cleanup found along the way.
+
+ Resolves: #39
+
+#### res_pjsip: mediasec: Add Security-Client headers after 401
+ Author: Maximilian Fridrich
+ Date: 2023-05-02
+
+ When using mediasec, requests sent after a 401 must still contain the
+ Security-Client header according to
+ draft-dawes-sipcore-mediasec-parameter.
+
+ Resolves: #48
+
+#### LICENSE: Update link to trademark policy.
+ Author: Joshua C. Colp
+ Date: 2023-05-01
+
+ Resolves: #43
+
+#### chan_dahdi: Add dialmode option for FXS lines.
+ Author: Naveen Albert
+ Date: 2023-04-28
+
+ Currently, both pulse and tone dialing are always enabled
+ on all FXS lines, with no way of disabling one or the other.
+
+ In some circumstances, it is desirable or necessary to
+ disable one of these, and this behavior can be problematic.
+
+ A new "dialmode" option is added which allows setting the
+ methods to support on a per channel basis for FXS (FXO
+ signalled lines). The four options are "both", "pulse",
+ "dtmf"/"tone", and "none".
+
+ Additionally, integration with the CHANNEL function is
+ added so that this setting can be updated for a channel
+ during a call.
+
+ Resolves: #35
+ ASTERISK-29992
+
+ UserNote: A "dialmode" option has been added which allows
+ specifying, on a per-channel basis, what methods of
+ subscriber dialing (pulse and/or tone) are permitted.
+
+ Additionally, this can be changed on a channel
+ at any point during a call using the CHANNEL
+ function.
+
+
+#### Initial GitHub PRs
+ Author: George Joseph
+ Date: 2023-04-28
+
+
+#### Initial GitHub Issue Templates
+ Author: George Joseph
+ Date: 2023-04-28
+
+
+#### pbx_dundi: Fix PJSIP endpoint configuration check.
+ Author: Joshua C. Colp
+ Date: 2023-04-13
+
+ ASTERISK-28233
+
+
+#### Revert "app_queue: periodic announcement configurable start time."
+ Author: Joshua Colp
+ Date: 2023-04-11
+
+ This reverts commit 3fd0b65bae4b1b14434737ffcf0da4aa9ff717f6.
+
+ Reason for revert: Causes segmentation fault.
+
+
+#### res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
+ Author: Naveen Albert
+ Date: 2023-02-17
+
+ The current STIR/SHAKEN signing process is inconsistent with the
+ RFCs in a couple ways that can cause interoperability issues.
+
+ RFC8225 specifies that the keys must be ordered lexicographically, but
+ currently the fields are simply ordered according to the order
+ in which they were added to the JSON object, which is not
+ compliant with the RFC and can cause issues with some carriers.
+
+ To fix this, we now leverage libjansson's ability to dump a JSON
+ object sorted by key value, yielding the correct field ordering.
+
+ Additionally, telephone numbers must have any leading + prefix removed
+ and must not contain characters outside of 0-9, *, and # in order
+ to comply with the RFCs. Numbers are now properly formatted as such.
+
+ ASTERISK-30407 #close
+
+
+#### pbx_dundi: Add PJSIP support.
+ Author: Naveen Albert
+ Date: 2022-12-09
+
+ Adds PJSIP as a supported technology to DUNDi.
+
+ To facilitate this, we now allow an endpoint to be specified
+ for outgoing PJSIP calls. We also allow users to force a specific
+ channel technology for outgoing SIP-protocol calls.
+
+ ASTERISK-28109 #close
+ ASTERISK-28233 #close
+
+
+#### install_prereq: Add Linux Mint support.
+ Author: The_Blode
+ Date: 2023-03-17
+
+ ASTERISK-30359 #close
+
+
+#### chan_pjsip: fix music on hold continues after INVITE with replaces
+ Author: Henning Westerholt
+ Date: 2023-03-21
+
+ In a three party scenario with INVITE with replaces, we need to
+ unhold the call, otherwise one party continues to get music on
+ hold, and the call is not properly bridged between them.
+
+ ASTERISK-30428
+
+
+#### voicemail.conf: Fix incorrect comment about #include.
+ Author: Naveen Albert
+ Date: 2023-03-28
+
+ A comment at the top of voicemail.conf says that #include
+ cannot be used in voicemail.conf because this breaks
+ the ability for app_voicemail to auto-update passwords.
+ This is factually incorrect, since Asterisk has no problem
+ updating files that are #include'd in the main configuration
+ file, and this does work in voicemail.conf as well.
+
+ ASTERISK-30479 #close
+
+
+#### app_queue: Fix minor xmldoc duplication and vagueness.
+ Author: Naveen Albert
+ Date: 2023-04-03
+
+ The F option in the xmldocs for the Queue application
+ was erroneously duplicated, causing it to display
+ twice on the wiki. The two sections are now merged into one.
+
+ Additionally, the description for the d option was quite
+ vague. Some more details are added to provide context
+ as to what this actually does.
+
+ ASTERISK-30486 #close
+
+
+#### test.c: Fix counting of tests and add 2 new tests
+ Author: George Joseph
+ Date: 2023-03-28
+
+ The unit test XML output was counting all registered tests as "run"
+ even when only a subset were actually requested to be run and
+ the "failures" attribute was missing.
+
+ * The "tests" attribute of the "testsuite" element in the
+ output XML now reflects only the tests actually requested
+ to be executed instead of all the tests registered.
+
+ * The "failures" attribute was added to the "testsuite"
+ element.
+
+ Also added 2 new unit tests that just pass and fail to be
+ used for CI testing.
+
+
+#### res_calendar: output busy state as part of show calendar.
+ Author: Jaco Kroon
+ Date: 2023-03-23
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### loader.c: Minor module key check simplification.
+ Author: Sean Bright
+ Date: 2023-03-23
+
+
+#### ael: Regenerate lexers and parsers.
+ Author: Sean Bright
+ Date: 2023-03-21
+
+ Various changes to ensure that the lexers and parsers can be correctly
+ generated when REBUILD_PARSERS is enabled.
+
+ Some notes:
+
+ * Because of the version of flex we are using to generate the lexers
+ (2.5.35) some post-processing in the Makefile is still required.
+
+ * The generated lexers do not contain the problematic C99 check that
+ was being replaced by the call to sed in the respective Makefiles so
+ it was removed.
+
+ * Since these files are generated, they will include trailing
+ whitespace in some places. This does not need to be corrected.
+
+
+#### bridge_builtin_features: add beep via touch variable
+ Author: Mike Bradeen
+ Date: 2023-03-01
+
+ Add periodic beep option to one-touch recording by setting
+ the touch variable TOUCH_MONITOR_BEEP or
+ TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
+
+ If the interval is less than 5 seconds, a minimum of 5
+ seconds will be imposed. If the interval is set to an
+ invalid value, it will default to 15 seconds.
+
+ A new test event PERIODIC_HOOK_ENABLED was added to the
+ func_periodic_hook hook_on function to indicate when
+ a hook is started. This is so we can test that the touch
+ variable starts the hook as expected.
+
+ ASTERISK-30446
+
+
+#### res_mixmonitor: MixMonitorMute by MixMonitor ID
+ Author: Mike Bradeen
+ Date: 2023-03-13
+
+ While it is possible to create multiple mixmonitor instances
+ on a channel, it was not previously possible to mute individual
+ instances.
+
+ This change includes the ability to specify the MixMonitorID
+ when calling the manager action: MixMonitorMute. This will
+ allow an individual MixMonitor instance to be muted via id.
+ This id can be stored as a channel variable using the 'i'
+ MixMonitor option.
+
+ As part of this change, if no MixMonitorID is specified in
+ the manager action MixMonitorMute, Asterisk will set the mute
+ flag on all MixMonitor spy-type audiohooks on the channel.
+ This is done via the new audiohook function:
+ ast_audiohook_set_mute_all.
+
+ ASTERISK-30464
+
+
+#### format_sln: add .slin as supported file extension
+ Author: Mike Bradeen
+ Date: 2023-03-14
+
+ Adds '.slin' to existing supported file extensions:
+ .sln and .raw
+
+ ASTERISK-30465
+
+
+#### res_agi: RECORD FILE plays 2 beeps.
+ Author: Sean Bright
+ Date: 2023-03-08
+
+ Sending the "RECORD FILE" command without the optional
+ `offset_samples` argument can result in two beeps playing on the
+ channel.
+
+ This bug has been present since Asterisk 0.3.0 (2003-02-06).
+
+ ASTERISK-30457 #close
+
+
+#### func_json: Fix JSON parsing issues.
+ Author: Naveen Albert
+ Date: 2023-02-26
+
+ Fix issue with returning empty instead of dumping
+ the JSON string when recursing.
+
+ Also adds a unit test to capture this fix.
+
+ ASTERISK-30441 #close
+
+
+#### app_senddtmf: Add SendFlash AMI action.
+ Author: Naveen Albert
+ Date: 2023-02-26
+
+ Adds an AMI action to send a flash event
+ on a channel.
+
+ ASTERISK-30440 #close
+
+
+#### app_dial: Fix DTMF not relayed to caller on unanswered calls.
+ Author: Naveen Albert
+ Date: 2023-03-04
+
+ DTMF frames are not handled in app_dial when sent towards the
+ caller. This means that if DTMF is sent to the calling party
+ and the call has not yet been answered, the DTMF is not audible.
+ This is now fixed by relaying DTMF frames if only a single
+ destination is being dialed.
+
+ ASTERISK-29516 #close
+
+
+#### configure: fix detection of re-entrant resolver functions
+ Author: Fabrice Fontaine
+ Date: 2023-03-08
+
+ uClibc does not provide res_nsearch:
+ asterisk-16.0.0/main/dns.c:506: undefined reference to `res_nsearch'
+
+ Patch coded by Yann E. MORIN:
+ http://lists.busybox.net/pipermail/buildroot/2018-October/232630.html
+
+ ASTERISK-21795 #close
+
+ Signed-off-by: Bernd Kuhls <bernd.kuhls@t-online.de>
+ [Retrieved from:
+ https: //git.buildroot.net/buildroot/tree/package/asterisk/0005-configure-fix-detection-of-re-entrant-resolver-funct.patch]
+ Signed-off-by: Fabrice Fontaine <fontaine.fabrice@gmail.com>
+
+#### cli: increase channel column width
+ Author: Mike Bradeen
+ Date: 2023-03-06
+
+ For 'core show channels', the Channel name field is increased
+ to 64 characters and the Location name field is increased to
+ 32 characters.
+
+ For 'core show channels verbose', the Channel name field is
+ increased to 80 characters, the Context is increased to 24
+ characters and the Extension is increased to 24 characters.
+
+ ASTERISK-30455
+
+
+#### app_queue: periodic announcement configurable start time.
+ Author: Jaco Kroon
+ Date: 2023-02-21
+
+ This newly introduced periodic-announce-startdelay makes it possible to
+ configure the initial start delay of the first periodic announcement
+ after which periodic-announce-frequency takes over.
+
+ ASTERISK-30437 #close
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### make_version: Strip svn stuff and suppress ref HEAD errors
+ Author: George Joseph
+ Date: 2023-03-13
+
+ * All of the code that used subversion has been removed.
+
+ * When Asterisk is checked out from a tag or commit instead
+ of one of the regular branches, git would emit messages like
+ "fatal: ref HEAD is not a symbolic ref" which weren't fatal
+ at all. Those are now suppressed.
+
+
+#### res_http_media_cache: Introduce options and customize
+ Author: Holger Hans Peter Freyther
+ Date: 2022-10-16
+
+ Make the existing CURL parameters configurable and allow
+ to specify the usable protocols, proxy and DNS timeout.
+
+ ASTERISK-30340
+
+
+#### main/iostream.c: fix build with libressl
+ Author: Fabrice Fontaine
+ Date: 2023-02-25
+
+ Fix the following build failure with libressl by using SSL_is_server
+ which is available since version 2.7.0 and
+ https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:
+
+ iostream.c: In function 'ast_iostream_close':
+ iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
+ 559 | if (!stream->ssl->server) {
+ | ^~
+
+ ASTERISK-30107 #close
+
+ Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
+
+#### contrib: rc.archlinux.asterisk uses invalid redirect.
+ Author: Sean Bright
+ Date: 2023-03-02
+
+ `rc.archlinux.asterisk`, which explicitly requests bash in its
+ shebang, uses the following command syntax:
+
+ ${DAEMON} -rx "core stop now" > /dev/null 2&>1
+
+ The intent of which is to execute:
+
+ ${DAEMON} -rx "core stop now"
+
+ While sending both stdout and stderr to `/dev/null`. Unfortunately,
+ because the `&` is in the wrong place, bash is interpreting the `2` as
+ just an additional argument to the `$DAEMON` command and not as a file
+ descriptor and proceeds to use the bashism `&>` to send stderr and
+ stdout to a file named `1`.
+
+ So we clean it up and just use bash's shortcut syntax.
+
+ Issue raised and a fix suggested (but not used) by peutch on GitHub¹.
+
+ ASTERISK-30449 #close
+
+ 1. https://github.com/asterisk/asterisk/pull/31
+
+
+#### res_pjsip_pubsub: subscription cleanup changes
+ Author: Mike Bradeen
+ Date: 2023-03-29
+
+ There are two main parts of the change associated with this
+ commit. These are driven by the change in call order of
+ pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
+ when an in-dialog SUBSCRIBE is received.
+
+ First, the previous behavior was for pjproject to call
+ pubsub_on_rx_refresh before calling pubsub_on_evsub_state
+ when an in-dialog SUBSCRIBE was received that changes the
+ subscription state.
+
+ If that change was a termination due to a re-SUBSCRIBE with
+ an expires of 0, we used to use the call to pubsub_on_rx_refresh
+ to set the substate of the evsub to TERMINATE_PENDING before
+ pjproject could call pubsub_on_evsub_state.
+
+ This substate let pubsub_on_evsub_state know that the
+ subscription TERMINATED event could be ignored as there was
+ still a subsequent NOTIFY that needed to be generated and
+ another call to pubsub_on_evsub_state to come with it.
+
+ That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
+ which would see the TERMINATE_PENDING state and transition it
+ to TERMINATE_IN_PROGRESS before triggering another call to
+ pubsub_on_evsub_state (which now would clean up the evsub.)
+
+ The new pjproject behavior is to call pubsub_on_evsub_state
+ before pubsub_on_rx_refresh. This means we no longer can set
+ the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
+ that it can ignore the first TERMINATED event.
+
+ To handle this, we now look directly at the event type,
+ method type and the expires value to determine whether we
+ want to ignore the event or use it to trigger the evsub
+ cleanup.
+
+ Second, pjproject now expects the NOTIFY to actually be sent
+ during pubsub_on_rx_refresh and avoids the protocol violation
+ inherent in sending a NOTIFY before the SUBSCRIBE is
+ acknowledged by caching the sent NOTIFY then sending it
+ after responding to the SUBSCRIBE.
+
+ This requires we send the NOTIFY using the non-serialized
+ pubsub_on_refresh_timeout directly and let pjproject handle
+ the protocol violation.
+
+ ASTERISK-30469
+
+
+#### Revert "pbx_ael: Global variables are not expanded."
+ Author: Sean Bright
+ Date: 2023-03-19
+
+ This reverts commit 56051d1ac5115ff8c55b920fc441613c487fb512.
+
+ Reason for revert: Behavior change that breaks existing dialplan.
+
+ ASTERISK-30472 #close
+
+
+#### res_pjsip: Replace invalid UTF-8 sequences in callerid name
+ Author: George Joseph
+ Date: 2023-02-16
+
+ * Added a new function ast_utf8_replace_invalid_chars() to
+ utf8.c that copies a string replacing any invalid UTF-8
+ sequences with the Unicode specified U+FFFD replacement
+ character. For example: "abc\xffdef" becomes "abc\uFFFDdef".
+ Any UTF-8 compliant implementation will show that character
+ as a � character.
+
+ * Updated res_pjsip:set_id_from_hdr() to use
+ ast_utf8_replace_invalid_chars and print a warning if any
+ invalid sequences were found during the copy.
+
+ * Updated stasis_channels:ast_channel_publish_varset to use
+ ast_utf8_replace_invalid_chars and print a warning if any
+ invalid sequences were found during the copy.
+
+ ASTERISK-27830
+
+
+#### test.c: Avoid passing -1 to FD_* family of functions.
+ Author: Sean Bright
+ Date: 2023-02-27
+
+ This avoids buffer overflow errors when running tests that capture
+ output from child processes.
+
+ This also corrects a copypasta in an off-nominal error message.
+
+
+#### chan_iax2: Fix jitterbuffer regression prior to receiving audio.
+ Author: Naveen Albert
+ Date: 2022-12-14
+
+ ASTERISK_29392 (a security fix) introduced a regression by
+ not processing frames when we don't have an audio format.
+
+ Currently, chan_iax2 only calls jb_get to read frames from
+ the jitterbuffer when the voiceformat has been set on the pvt.
+ However, this only happens when we receive a voice frame, which
+ means that prior to receiving voice frames, other types of frames
+ get stalled completely in the jitterbuffer.
+
+ To fix this, we now fallback to using the format negotiated during
+ call setup until we've actually received a voice frame with a format.
+ This ensures we're always able to read from the jitterbuffer.
+
+ ASTERISK-30354 #close
+ ASTERISK-30162 #close
+
+
+#### test_crypto.c: Fix getcwd(…) build error.
+ Author: Sean Bright
+ Date: 2023-02-27
+
+ `getcwd(…)` is decorated with the `warn_unused_result` attribute and
+ therefore needs its return value checked.
+
+
+#### pjproject_bundled: Fix cross-compilation with SSL libs.
+ Author: Nick French
+ Date: 2023-02-11
+
+ Asterisk makefiles auto-detect SSL library availability,
+ then they assume that pjproject makefiles will also autodetect
+ an SSL library at the same time, so they do not pass on the
+ autodetection result to pjproject.
+
+ This normally works, except the pjproject makefiles disables
+ autodetection when cross-compiling.
+
+ Fix by explicitly configuring pjproject to use SSL if we
+ have been told to use it or it was autodetected
+
+ ASTERISK-30424 #close
+
+
+#### app_read: Add an option to return terminator on empty digits.
+ Author: Mike Bradeen
+ Date: 2023-01-30
+
+ Adds 'e' option to allow Read() to return the terminator as the
+ dialed digits in the case where only the terminator is entered.
+
+ ie; if "#" is entered, return "#" if the 'e' option is set and ""
+ if it is not.
+
+ ASTERISK-30411
+
+
+#### res_phoneprov.c: Multihomed SERVER cache prevention
+ Author: cmaj
+ Date: 2023-01-07
+
+ Phones moving between subnets on multi-homed server have their
+ initially connected interface IP cached in the SERVER variable,
+ even when it is not specified in the configuration files. This
+ prevents phones from obtaining the correct SERVER variable value
+ when they move to another subnet.
+
+ ASTERISK-30388 #close
+ Reported-by: cmaj
+
+
+#### app_directory: Add a 'skip call' option.
+ Author: Mike Bradeen
+ Date: 2023-01-27
+
+ Adds 's' option to skip calling the extension and instead set the
+ extension as DIRECTORY_EXTEN channel variable.
+
+ ASTERISK-30405
+
+
+#### app_senddtmf: Add option to answer target channel.
+ Author: Mike Bradeen
+ Date: 2023-02-06
+
+ Adds a new option to SendDTMF() which will answer the specified
+ channel if it is not already up. If no channel is specified, the
+ current channel will be answered instead.
+
+ ASTERISK-30422
+
+
+#### res_pjsip: Prevent SEGV in pjsip_evsub_send_request
+ Author: Mike Bradeen
+ Date: 2023-02-21
+
+ contributed pjproject - patch to check sub->pending_notify
+ in evsub.c:on_tsx_state before calling
+ pjsip_evsub_send_request()
+
+ res_pjsip_pubsub - change post pjsip 2.13 behavior to use
+ pubsub_on_refresh_timeout to avoid the ao2_cleanup call on
+ the sub_tree. This is is because the final NOTIFY send is no
+ longer the last place the sub_tree is referenced.
+
+ ASTERISK-30419
+
+
+#### app_queue: Minor docs and logging fixes for UnpauseQueueMember.
+ Author: Sean Bright
+ Date: 2023-02-02
+
+ ASTERISK-30417 #close
+
+
+#### app_queue: Reset all queue defaults before reload.
+ Author: Sean Bright
+ Date: 2023-01-31
+
+ Several queue fields were not being set to their default value during
+ a reload.
+
+ Additionally added some sample configuration options that were missing
+ from queues.conf.sample.
+
+
+#### res_pjsip: Upgraded bundled pjsip to 2.13
+ Author: Mike Bradeen
+ Date: 2023-01-20
+
+ Removed multiple patches.
+
+ Code chages in res_pjsip_pubsub due to changes in evsub.
+
+ Pjsip now calls on_evsub_state() before on_rx_refresh(),
+ so the sub tree deletion that used to take place in
+ on_evsub_state() now must take place in on_rx_refresh().
+
+ Additionally, pjsip now requires that you send the NOTIFY
+ from within on_rx_refresh(), otherwise it will assert
+ when going to send the 200 OK. The idea is that it will
+ look for this NOTIFY and cache it until after sending the
+ response in order to deal with the self-imposed message
+ mis-order. Asterisk previously dealt with this by pushing
+ the NOTIFY in on_rx_refresh(), but pjsip now forces us
+ to use it's method.
+
+ Changes were required to configure in order to detect
+ which way pjsip handles this as the two are not
+ compatible for the reasons mentioned above.
+
+ A corresponding change in testsuite is required in order
+ to deal with the small interal timing changes caused by
+ moving the NOTIFY send.
+
+ ASTERISK-30325
+
+
+#### doxygen: Fix doxygen errors.
+ Author: Sean Bright
+ Date: 2023-01-30
+
+
+#### app_signal: Add signaling applications
+ Author: Naveen Albert
+ Date: 2022-01-06
+
+ Adds the Signal and WaitForSignal
+ applications, which can be used for inter-channel
+ signaling in the dialplan.
+
+ Signal supports sending a signal to other channels
+ listening for a signal of the same name, with an
+ optional data payload. The signal is received by
+ all channels waiting for that named signal.
+
+ ASTERISK-29810 #close
+
+
+#### app_directory: add ability to specify configuration file
+ Author: Mike Bradeen
+ Date: 2023-01-25
+
+ Adds option to app_directory to specify a filename from which to
+ read configuration instead of voicemail.conf ie;
+
+ same => n,Directory(,,c(directory.conf))
+
+ This configuration should contain a list of extensions using the
+ voicemail.conf format, ie;
+
+ 2020=2020,Dog Dog,,,,attach=no|saycid=no|envelope=no|delete=no
+
+ ASTERISK-30404
+
+
+#### func_json: Enhance parsing capabilities of JSON_DECODE
+ Author: Naveen Albert
+ Date: 2022-02-12
+
+ Adds support for arrays to JSON_DECODE by allowing the
+ user to print out entire arrays or index a particular
+ key or print the number of keys in a JSON array.
+
+ Additionally, adds support for recursively iterating a
+ JSON tree in a single function call, making it easier
+ to parse JSON results with multiple levels. A maximum
+ depth is imposed to prevent potentially blowing
+ the stack.
+
+ Also fixes a bug with the unit tests causing an empty
+ string to be printed instead of the actual test result.
+
+ ASTERISK-29913 #close
+
+
+#### res_stasis_snoop: Fix snoop crash
+ Author: sungtae kim
+ Date: 2023-01-04
+
+ Added NULL pointer check and channel lock to prevent resource release
+ while the chanspy is processing.
+
+ ASTERISK-29604
+
+
+#### pbx_ael: Global variables are not expanded.
+ Author: Sean Bright
+ Date: 2023-01-26
+
+ Variable references within global variable assignments are now
+ expanded rather than being included literally.
+
+ ASTERISK-30406 #close
+
+
+#### res_pjsip_session: Add overlap_context option.
+ Author: Naveen Albert
+ Date: 2022-10-13
+
+ Adds the overlap_context option, which can be used
+ to explicitly specify a context to use for overlap
+ dialing extension matches, rather than forcibly
+ using the context configured for the endpoint.
+
+ ASTERISK-30262 #close
+
+
+#### app_playback.c: Fix PLAYBACKSTATUS regression.
+ Author: Sean Bright
+ Date: 2023-01-05
+
+ In Asterisk 11, if a channel was redirected away during Playback(),
+ the PLAYBACKSTATUS variable would be set to SUCCESS. In Asterisk 12
+ (specifically commit 7d9871b3940fa50e85039aef6a8fb9870a7615b9) that
+ behavior was inadvertently changed and the same operation would result
+ in the PLAYBACKSTATUS variable being set to FAILED. The Asterisk 11
+ behavior has been restored.
+
+ Partial fix for ASTERISK~25661.
+
+
+#### res_rtp_asterisk: Don't use double math to generate timestamps
+ Author: George Joseph
+ Date: 2023-01-11
+
+ Rounding issues with double math were causing rtp timestamp
+ slips in outgoing packets. We're now back to integer math
+ and are getting no more slips.
+
+ ASTERISK-30391
+
+
+#### format_wav: replace ast_log(LOG_DEBUG, ...) by ast_debug(1, ...)
+ Author: Alexei Gradinari
+ Date: 2023-01-06
+
+ Each playback of WAV files results in logging
+ "Skipping unknown block 'LIST'".
+
+ To prevent unnecessary flooding of this DEBUG log this patch replaces
+ ast_log(LOG_DEBUG, ...) by ast_debug(1, ...).
+
+
+#### res_pjsip_rfc3326: Add SIP causes support for RFC3326
+ Author: Igor Goncharovsky
+ Date: 2022-11-18
+
+ Add ability to set HANGUPCAUSE when SIP causecode received in BYE (in addition to currently supported Q.850).
+
+ ASTERISK-30319 #close
+
+
+#### res_rtp_asterisk: Asterisk Media Experience Score (MES)
+ Author: George Joseph
+ Date: 2022-10-28
+
+ -----------------
+
+ This commit reinstates MES with some casting fixes to the
+ functions in time.h that convert between doubles and timeval
+ structures. The casting issues were causing incorrect
+ timestamps to be calculated which caused transcoding from/to
+ G722 to produce bad or no audio.
+
+ ASTERISK-30391
+
+ -----------------
+
+ This module has been updated to provide additional
+ quality statistics in the form of an Asterisk
+ Media Experience Score. The score is avilable using
+ the same mechanisms you'd use to retrieve jitter, loss,
+ and rtt statistics. For more information about the
+ score and how to retrieve it, see
+ https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
+
+ * Updated chan_pjsip to set quality channel variables when a
+ call ends.
+ * Updated channels/pjsip/dialplan_functions.c to add the ability
+ to retrieve the MES along with the existing rtcp stats when
+ using the CHANNEL dialplan function.
+ * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
+ checks for debugging purposes.
+ * Added several function to time.h for manipulating time-in-samples
+ and times represented as double seconds.
+ * Updated rtp_engine.c to pass through the MES when stats are
+ requested. Also debug output that dumps the stats when an
+ rtp instance is destroyed.
+ * Updated res_rtp_asterisk.c to implement the calculation of the
+ MES. In the process, also had to update the calculation of
+ jitter. Many debugging statements were also changed to be
+ more informative.
+ * Added a unit test for internal testing. The test should not be
+ run during normal operation and is disabled by default.
+
+
+#### Revert "res_rtp_asterisk: Asterisk Media Experience Score (MES)"
+ Author: George Joseph
+ Date: 2023-01-09
+
+ This reverts commit d454801c2ddba89f7925c847012db2866e271f68.
+
+ Reason for revert: Issue when transcoding to/from g722
+
+
+#### loader: Allow declined modules to be unloaded.
+ Author: Naveen Albert
+ Date: 2022-12-08
+
+ Currently, if a module declines to load, dlopen is called
+ to register the module but dlclose never gets called.
+ Furthermore, loader.c currently doesn't allow dlclose
+ to ever get called on the module, since it declined to
+ load and the unload function bails early in this case.
+
+ This can be problematic if a module is updated, since the
+ new module cannot be loaded into memory since we haven't
+ closed all references to it. To fix this, we now allow
+ modules to be unloaded, even if they never "loaded" in
+ Asterisk itself, so that dlclose is called and the module
+ can be properly cleaned up, allowing the updated module
+ to be loaded from scratch next time.
+
+ ASTERISK-30345 #close
+
+
+#### app_broadcast: Add Broadcast application
+ Author: Naveen Albert
+ Date: 2022-08-15
+
+ Adds a new application, Broadcast, which can be used for
+ one-to-many transmission and many-to-one reception of
+ channel audio in Asterisk. This is similar to ChanSpy,
+ except it is designed for multiple channel targets instead
+ of a single one. This can make certain kinds of audio
+ manipulation more efficient and streamlined. New kinds
+ of audio injection impossible with ChanSpy are also made
+ possible.
+
+ ASTERISK-30180 #close
+
+
+#### func_frame_trace: Print text for text frames.
+ Author: Naveen Albert
+ Date: 2022-12-13
+
+ Since text frames contain a text body, make FRAME_TRACE
+ more useful for text frames by actually printing the text.
+
+ ASTERISK-30353 #close
+
+
+#### json.h: Add ast_json_object_real_get.
+ Author: Naveen Albert
+ Date: 2022-12-16
+
+ json.h contains macros to get a string and an integer
+ from a JSON object. However, the macro to do this for
+ JSON reals is missing. This adds that.
+
+ ASTERISK-30361 #close
+
+
+#### manager: Fix appending variables.
+ Author: Naveen Albert
+ Date: 2022-12-22
+
+ The if statement here is always false after the for
+ loop finishes, so variables are never appended.
+ This removes that to properly append to the end
+ of the variable list.
+
+ ASTERISK-30351 #close
+ Reported by: Sebastian Gutierrez
+
+
+#### res_pjsip_transport_websocket: Add remote port to transport
+ Author: George Joseph
+ Date: 2022-12-23
+
+ When Asterisk receives a new websocket conenction, it creates a new
+ pjsip transport for it and copies connection data into it. The
+ transport manager then uses the remote IP address and port on the
+ transport to create a monitor for each connection. However, the
+ remote port wasn't being copied, only the IP address which meant
+ that the transport manager was creating only 1 monitoring entry for
+ all websocket connections from the same IP address. Therefore, if
+ one of those connections failed, it deleted the transport taking
+ all the the connections from that same IP address with it.
+
+ * We now copy the remote port into the created transport and the
+ transport manager behaves correctly.
+
+ ASTERISK-30369
+
+
+#### http.c: Fix NULL pointer dereference bug
+ Author: Boris P. Korzun
+ Date: 2022-12-28
+
+ If native HTTP is disabled but HTTPS is enabled and status page enabled
+ too, Core/HTTP crashes while loading. 'global_http_server' references
+ to NULL, but the status page tries to dereference it.
+
+ The patch adds a check for HTTP is enabled.
+
+ ASTERISK-30379 #close
+
+
+#### res_http_media_cache: Do not crash when there is no extension
+ Author: Holger Hans Peter Freyther
+ Date: 2022-12-16
+
+ Do not crash when a URL has no path component as in this case the
+ ast_uri_path function will return NULL. Make the code cope with not
+ having a path.
+
+ The below would crash
+ > media cache create http://google.com /tmp/foo.wav
+
+ Thread 1 "asterisk" received signal SIGSEGV, Segmentation fault.
+ 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
+ (gdb) bt
+ #0 0x0000ffff836616cc in strrchr () from /lib/aarch64-linux-gnu/libc.so.6
+ #1 0x0000ffff43d43a78 in file_extension_from_string (str=<optimized out>, buffer=buffer@entry=0xffffca9973c0 "",
+ capacity=capacity@entry=64) at res_http_media_cache.c:288
+ #2 0x0000ffff43d43bac in file_extension_from_url_path (bucket_file=bucket_file@entry=0x3bf96568,
+ buffer=buffer@entry=0xffffca9973c0 "", capacity=capacity@entry=64) at res_http_media_cache.c:378
+ #3 0x0000ffff43d43c74 in bucket_file_set_extension (bucket_file=bucket_file@entry=0x3bf96568) at res_http_media_cache.c:392
+ #4 0x0000ffff43d43d10 in bucket_file_run_curl (bucket_file=0x3bf96568) at res_http_media_cache.c:555
+ #5 0x0000ffff43d43f74 in bucket_http_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
+ at res_http_media_cache.c:613
+ #6 0x0000000000487638 in bucket_file_wizard_create (sorcery=<optimized out>, data=<optimized out>, object=<optimized out>)
+ at bucket.c:191
+ #7 0x0000000000554408 in sorcery_wizard_create (object_wizard=object_wizard@entry=0x3b9f0718,
+ details=details@entry=0xffffca9974a8) at sorcery.c:2027
+ #8 0x0000000000559698 in ast_sorcery_create (sorcery=<optimized out>, object=object@entry=0x3bf96568) at sorcery.c:2077
+ #9 0x00000000004893a4 in ast_bucket_file_create (file=file@entry=0x3bf96568) at bucket.c:727
+ #10 0x00000000004f877c in ast_media_cache_create_or_update (uri=0x3bfa1103 "https://google.com",
+ file_path=0x3bfa1116 "/tmp/foo.wav", metadata=metadata@entry=0x0) at media_cache.c:335
+ #11 0x00000000004f88ec in media_cache_handle_create_item (e=<optimized out>, cmd=<optimized out>, a=0xffffca9976b8)
+ at media_cache.c:640
+
+ ASTERISK-30375 #close
+
+
+#### res_rtp_asterisk: Asterisk Media Experience Score (MES)
+ Author: George Joseph
+ Date: 2022-10-28
+
+ This module has been updated to provide additional
+ quality statistics in the form of an Asterisk
+ Media Experience Score. The score is avilable using
+ the same mechanisms you'd use to retrieve jitter, loss,
+ and rtt statistics. For more information about the
+ score and how to retrieve it, see
+ https://wiki.asterisk.org/wiki/display/AST/Media+Experience+Score
+
+ * Updated chan_pjsip to set quality channel variables when a
+ call ends.
+ * Updated channels/pjsip/dialplan_functions.c to add the ability
+ to retrieve the MES along with the existing rtcp stats when
+ using the CHANNEL dialplan function.
+ * Added the ast_debug_rtp_is_allowed and ast_debug_rtcp_is_allowed
+ checks for debugging purposes.
+ * Added several function to time.h for manipulating time-in-samples
+ and times represented as double seconds.
+ * Updated rtp_engine.c to pass through the MES when stats are
+ requested. Also debug output that dumps the stats when an
+ rtp instance is destroyed.
+ * Updated res_rtp_asterisk.c to implement the calculation of the
+ MES. In the process, also had to update the calculation of
+ jitter. Many debugging statements were also changed to be
+ more informative.
+ * Added a unit test for internal testing. The test should not be
+ run during normal operation and is disabled by default.
+
+ ASTERISK-30280
+
+
+#### pbx_app: Update outdated pbx_exec channel snapshots.
+ Author: Naveen Albert
+ Date: 2022-12-21
+
+ pbx_exec makes a channel snapshot before executing applications.
+ This doesn't cause an issue during normal dialplan execution
+ where pbx_exec is called over and over again in succession.
+ However, if pbx_exec is called "one off", e.g. using
+ ast_pbx_exec_application, then a channel snapshot never ends
+ up getting made after the executed application returns, and
+ inaccurate snapshot information will linger for a while, causing
+ "core show channels", etc. to show erroneous info.
+
+ This is fixed by manually making a channel snapshot at the end
+ of ast_pbx_exec_application, since we anticipate that pbx_exec
+ might not get called again immediately.
+
+ ASTERISK-30367 #close
+
+
+#### res_pjsip_session: Use Caller ID for extension matching.
+ Author: Naveen Albert
+ Date: 2022-11-26
+
+ Currently, there is no Caller ID available to us when
+ checking for an extension match when handling INVITEs.
+ As a result, extension patterns that depend on the Caller ID
+ are not matched and calls may be incorrectly rejected.
+
+ The Caller ID is not available because the supplement that
+ adds Caller ID to the session does not execute until after
+ this check. Supplement callbacks cannot yet be executed
+ at this point since the session is not yet in the appropriate
+ state.
+
+ To fix this without impacting existing behavior, the Caller ID
+ number is now retrieved before attempting to pattern match.
+ This ensures pattern matching works correctly and there is
+ no behavior change to the way supplements are called.
+
+ ASTERISK-28767 #close
+
+
+#### res_pjsip_sdp_rtp.c: Use correct timeout when put on hold.
+ Author: Ben Ford
+ Date: 2022-12-12
+
+ When a call is put on hold and it has moh_passthrough and rtp_timeout
+ set on the endpoint, the wrong timeout will be used. rtp_timeout_hold is
+ expected to be used, but rtp_timeout is used instead. This change adds a
+ couple of checks for locally_held to determine if rtp_timeout_hold needs
+ to be used instead of rtp_timeout.
+
+ ASTERISK-30350
+
+
+#### app_voicemail_odbc: Fix string overflow warning.
+ Author: Naveen Albert
+ Date: 2022-11-14
+
+ Fixes a negative offset warning by initializing
+ the buffer to empty.
+
+ Additionally, although it doesn't currently complain
+ about it, the size of a buffer is increased to
+ accomodate the maximum size contents it could have.
+
+ ASTERISK-30240 #close
+
+
+#### func_callerid: Warn about invalid redirecting reason.
+ Author: Naveen Albert
+ Date: 2022-11-26
+
+ Currently, if a user attempts to set a Caller ID related
+ function to an invalid value, a warning is emitted,
+ except for when setting the redirecting reason.
+ We now emit a warning if we were unable to successfully
+ parse the user-provided reason.
+
+ ASTERISK-30332 #close
+
+
+#### res_pjsip: Fix path usage in case dialing with '@'
+ Author: Igor Goncharovsky
+ Date: 2022-11-04
+
+ Fix aor lookup on sip path addition. Issue happens in case of dialing
+ with @ and overriding user part of RURI.
+
+ ASTERISK-30100 #close
+ Reported-by: Yury Kirsanov
+
+
+#### streams: Ensure that stream is closed in ast_stream_and_wait on error
+ Author: Peter Fern
+ Date: 2022-11-22
+
+ When ast_stream_and_wait returns an error (for example, when attempting
+ to stream to a channel after hangup) the stream is not closed, and
+ callers typically do not check the return code. This results in leaking
+ file descriptors, leading to resource exhaustion.
+
+ This change ensures that the stream is closed in case of error.
+
+ ASTERISK-30198 #close
+ Reported-by: Julien Alie
+
+
+#### app_sendtext: Remove references to removed applications.
+ Author: Naveen Albert
+ Date: 2022-12-10
+
+ Removes see-also references to applications that don't
+ exist anymore (removed in Asterisk 19),
+ so these dead links don't show up on the wiki.
+
+ ASTERISK-30347 #close
+
+
+#### res_geoloc: fix NULL pointer dereference bug
+ Author: Alexandre Fournier
+ Date: 2022-12-09
+
+ The `ast_geoloc_datastore_add_eprofile` function does not return 0 on
+ success, it returns the size of the underlying datastore. This means
+ that the datastore will be freed and its pointer set to NULL when no
+ error occured at all.
+
+ ASTERISK-30346
+
+
+#### res_pjsip_aoc: Don't assume a body exists on responses.
+ Author: Joshua C. Colp
+ Date: 2022-12-13
+
+ When adding AOC to an outgoing response the code
+ assumed that a body would exist for comparing the
+ Content-Type. This isn't always true.
+
+ The code now checks to make sure the response has
+ a body before checking the Content-Type.
+
+ ASTERISK-21502
+
+
+#### app_if: Fix format truncation errors.
+ Author: Naveen Albert
+ Date: 2022-12-12
+
+ Fixes format truncation warnings in gcc 12.2.1.
+
+ ASTERISK-30349 #close
+
+
+#### manager: AOC-S support for AOCMessage
+ Author: Michael Kuron
+ Date: 2022-11-01
+
+ ASTERISK-21502
+
+
+#### res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip
+ Author: Michael Kuron
+ Date: 2022-10-23
+
+ chan_sip supported sending AOC-D and AOC-E information in SIP INFO
+ messages in an "AOC" header in a format that was originally defined by
+ Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC
+ format that is supported by devices from multiple vendors, including
+ Snom phones with firmware >= 8.4.2 (released in 2010).
+
+ This commit adds a new res_pjsip_aoc module that inserts AOC information
+ into outgoing messages or sends SIP INFO messages as described below.
+ It also fixes a small issue in res_pjsip_session which didn't always
+ call session supplements on outgoing_response.
+
+ * AOC-S in the 180/183/200 responses to an INVITE request
+ * AOC-S in SIP INFO (if a 200 response has already been sent or if the
+ INVITE was sent by Asterisk)
+ * AOC-D in SIP INFO
+ * AOC-D in the 200 response to a BYE request (if the client hangs up)
+ * AOC-D in a BYE request (if Asterisk hangs up)
+ * AOC-E in the 200 response to a BYE request (if the client hangs up)
+ * AOC-E in a BYE request (if Asterisk hangs up)
+
+ The specification defines one more, AOC-S in an INVITE request, which
+ is not implemented here because it is not currently possible in
+ Asterisk to have AOC data ready at this point in call setup. Once
+ specifying AOC-S via the dialplan or passing it through from another
+ SIP channel's INVITE is possible, that might be added.
+
+ The SIP INFO requests are sent out immediately when the AOC indication
+ is received. The others are inserted into an appropriate outgoing
+ message whenever that is ready to be sent. In the latter case, the XML
+ is stored in a channel variable at the time the AOC indication is
+ received. Depending on where the AOC indications are coming from (e.g.
+ PRI or AMI), it may not always be possible to guarantee that the AOC-E
+ is available in time for the BYE.
+
+ Successfully tested AOC-D and both variants of AOC-E with a Snom D735
+ running firmware 10.1.127.10. It does not appear to properly support
+ AOC-S however, so that could only be tested by inspecting SIP traces.
+
+ ASTERISK-21502 #close
+ Reported-by: Matt Jordan <mjordan@digium.com>
+
+
+#### ari: Destroy body variables in channel create.
+ Author: Joshua C. Colp
+ Date: 2022-12-08
+
+ When passing a JSON body to the 'create' channel route
+ it would be converted into Asterisk variables, but never
+ freed resulting in a memory leak.
+
+ This change makes it so that the variables are freed in
+ all cases.
+
+ ASTERISK-30344
+
+
+#### app_voicemail: Fix missing email in msg_create_from_file.
+ Author: Naveen Albert
+ Date: 2022-11-03
+
+ msg_create_from_file currently does not dispatch emails,
+ which means that applications using this function, such
+ as MixMonitor, will not trigger notifications to users
+ (only AMI events are sent our currently). This is inconsistent
+ with other ways users can receive voicemail.
+
+ This is fixed by adding an option that attempts to send
+ an email and falling back to just the notifications as
+ done now if that fails. The existing behavior remains
+ the default.
+
+ ASTERISK-30283 #close
+
+
+#### res_pjsip: Fix typo in from_domain documentation
+ Author: Marcel Wagner
+ Date: 2022-11-25
+
+ This fixes a small typo in the from_domain documentation on the endpoint documentation
+
+ ASTERISK-30328 #close
+
+
+#### res_hep: Add support for named capture agents.
+ Author: Naveen Albert
+ Date: 2022-11-21
+
+ Adds support for the capture agent name field
+ of the Homer protocol to Asterisk by allowing
+ users to specify a name that will be sent to
+ the HEP server.
+
+ ASTERISK-30322 #close
+
+
+#### app_if: Adds conditional branch applications
+ Author: Naveen Albert
+ Date: 2021-06-28
+
+ Adds the If, ElseIf, Else, ExitIf, and EndIf
+ applications for conditional execution
+ of a block of dialplan, similar to the While,
+ EndWhile, and ExitWhile applications. The
+ appropriate branch is executed at most once
+ if available and may be broken out of while
+ inside.
+
+ ASTERISK-29497
+
+
+#### res_pjsip_session.c: Map empty extensions in INVITEs to s.
+ Author: Naveen Albert
+ Date: 2022-10-17
+
+ Some SIP devices use an empty extension for PLAR functionality.
+
+ Rather than rejecting these empty extensions, we now use the s
+ extension for such calls to mirror the existing PLAR functionality
+ in Asterisk (e.g. chan_dahdi).
+
+ ASTERISK-30265 #close
+
+
+#### res_pjsip: Update contact_user to point out default
+ Author: Marcel Wagner
+ Date: 2022-11-17
+
+ Updates the documentation for the 'contact_user' field to point out the
+ default outbound contact if no contact_user is specified 's'
+
+ ASTERISK-30316 #close
+
+
+#### res_adsi: Fix major regression caused by media format rearchitecture.
+ Author: Naveen Albert
+ Date: 2022-11-23
+
+ The commit that rearchitected media formats,
+ a2c912e9972c91973ea66902d217746133f96026 (ASTERISK_23114)
+ introduced a regression by improperly translating code in res_adsi.c.
+ In particular, the pointer to the frame buffer was initialized
+ at the top of adsi_careful_send, rather than dynamically updating it
+ for each frame, as is required.
+
+ This resulted in the first frame being repeatedly sent,
+ rather than advancing through the frames.
+ This corrupted the transmission of the CAS to the CPE,
+ which meant that CPE would never respond with the DTMF acknowledgment,
+ effectively completely breaking ADSI functionality.
+
+ This issue is now fixed, and ADSI now works properly again.
+
+ ASTERISK-29793 #close
+
+
+#### res_pjsip_header_funcs: Add custom parameter support.
+ Author: Naveen Albert
+ Date: 2022-07-21
+
+ Adds support for custom URI and header parameters
+ in the From header in PJSIP. Parameters can be
+ both set and read using this function.
+
+ ASTERISK-30150 #close
+
+
+#### func_presencestate: Fix invalid memory access.
+ Author: Naveen Albert
+ Date: 2022-11-13
+
+ When parsing information from AstDB while loading,
+ it is possible that certain pointers are never
+ set, which leads to invalid memory access and
+ then, fatally, invalid free attempts on this memory.
+ We now initialize to NULL to prevent this.
+
+ ASTERISK-30311 #close
+
+
+#### sig_analog: Fix no timeout duration.
+ Author: Naveen Albert
+ Date: 2022-12-01
+
+ ASTERISK_28702 previously attempted to fix an
+ issue with flash hook hold timing out after
+ just under 17 minutes, when it should have never
+ been timing out. It fixed this by changing 999999
+ to INT_MAX, but it did so in chan_dahdi, which
+ is the wrong place since ss_thread is now in
+ sig_analog and the one in chan_dahdi is mostly
+ dead code.
+
+ This fixes this by porting the fix to sig_analog.
+
+ ASTERISK-30336 #close
+
+
+#### xmldoc: Allow XML docs to be reloaded.
+ Author: Naveen Albert
+ Date: 2022-11-05
+
+ The XML docs are currently only loaded on
+ startup with no way to update them during runtime.
+ This makes it impossible to load modules that
+ use ACO/Sorcery (which require documentation)
+ if they are added to the source tree and built while
+ Asterisk is running (e.g. external modules).
+
+ This adds a CLI command to reload the XML docs
+ during runtime so that documentation can be updated
+ without a full restart of Asterisk.
+
+ ASTERISK-30289 #close
+
+
+#### rtp_engine.h: Update examples using ast_format_set.
+ Author: Naveen Albert
+ Date: 2022-11-24
+
+ This file includes some doxygen comments referencing
+ ast_format_set. This is an obsolete API that was
+ removed years back, but documentation was not fully
+ updated to reflect that. These examples are
+ updated to the current way of doing things
+ (using the format cache).
+
+ ASTERISK-30327 #close
+
+
+#### app_mixmonitor: Add option to use real Caller ID for voicemail.
+ Author: Naveen Albert
+ Date: 2022-11-04
+
+ MixMonitor currently uses the Connected Line as the Caller ID
+ for voicemails. This is due to the implementation being written
+ this way for use with Digium phones. However, in general this
+ is not correct for generic usage in the dialplan, and people
+ may need the real Caller ID instead. This adds an option to do that.
+
+ ASTERISK-30286 #close
+
+
+#### pjproject: 2.13 security fixes
+ Author: Ben Ford
+ Date: 2022-11-29
+
+ Backports two security fixes (c4d3498 and 450baca) from pjproject 2.13.
+
+ ASTERISK-30338
+
+
+#### pjsip_transport_events: Fix possible use after free on transport
+ Author: George Joseph
+ Date: 2022-10-10
+
+ It was possible for a module that registered for transport monitor
+ events to pass in a pjsip_transport that had already been freed.
+ This caused pjsip_transport_events to crash when looking up the
+ monitor for the transport. The fix is a two pronged approach.
+
+ 1. We now increment the reference count on pjsip_transports when we
+ create monitors for them, then decrement the count when the
+ transport is going to be destroyed.
+
+ 2. There are now APIs to register and unregister monitor callbacks
+ by "transport key" which is a string concatenation of the remote ip
+ address and port. This way the module needing to monitor the
+ transport doesn't have to hold on to the transport object itself to
+ unregister. It just has to save the transport_key.
+
+ * Added the pjsip_transport reference increment and decrement.
+
+ * Changed the internal transport monitor container key from the
+ transport->obj_name (which may not be unique anyway) to the
+ transport_key.
+
+ * Added a helper macro AST_SIP_MAKE_REMOTE_IPADDR_PORT_STR() that
+ fills a buffer with the transport_key using a passed-in
+ pjsip_transport.
+
+ * Added the following functions:
+ ast_sip_transport_monitor_register_key
+ ast_sip_transport_monitor_register_replace_key
+ ast_sip_transport_monitor_unregister_key
+ and marked their non-key counterparts as deprecated.
+
+ * Updated res_pjsip_pubsub and res_pjsip_outbound_register to use
+ the new "key" monitor functions.
+
+ NOTE: res_pjsip_registrar also uses the transport monitor
+ functionality but doesn't have a persistent object other than
+ contact to store a transport key. At this time, it continues to
+ use the non-key monitor functions.
+
+ ASTERISK-30244
+
+
+#### manager: prevent file access outside of config dir
+ Author: Mike Bradeen
+ Date: 2022-10-03
+
+ Add live_dangerously flag to manager and use this flag to
+ determine if a configuation file outside of AST_CONFIG_DIR
+ should be read.
+
+ ASTERISK-30176
+
+
+#### ooh323c: not checking for IE minimum length
+ Author: Mike Bradeen
+ Date: 2022-06-06
+
+ When decoding q.931 encoded calling/called number
+ now checking for length being less than minimum required.
+
+ ASTERISK-30103
+
+
+#### pbx_builtins: Allow Answer to return immediately.
+ Author: Naveen Albert
+ Date: 2022-11-11
+
+ The Answer application currently waits for up to 500ms
+ for media, even if users specify a different timeout.
+
+ This adds an option to not wait for media on the channel
+ by doing a raw answer instead. The default 500ms threshold
+ is also documented.
+
+ ASTERISK-30308 #close
+
+
+#### chan_dahdi: Allow FXO channels to start immediately.
+ Author: Naveen Albert
+ Date: 2022-11-11
+
+ Currently, chan_dahdi will wait for at least one
+ ring before an incoming call can enter the dialplan.
+ This is generally necessary in order to receive
+ the Caller ID spill and/or distinctive ringing
+ detection.
+
+ However, if neither of these is required, then there
+ is nothing gained by waiting for one ring and this
+ unnecessarily delays call setup. Users can now
+ use immediate=yes to make FXO channels (FXS signaled)
+ begin processing dialplan as soon as Asterisk receives
+ the call.
+
+ ASTERISK-30305 #close
+
+
+#### core & res_pjsip: Improve topology change handling.
+ Author: Maximilian Fridrich
+ Date: 2022-09-07
+
+ This PR contains two relatively separate changes in channel.c and
+ res_pjsip_session.c which ensure that topology changes are not ignored
+ in cases where they should be handled.
+
+ For channel.c:
+
+ The function ast_channel_request_stream_topology_change only triggers a
+ stream topology request change indication, if the channel's topology
+ does not equal the requested topology. However, a channel could be in a
+ state where it is currently "negotiating" a new topology but hasn't
+ updated it yet, so the topology request change would be lost. Channels
+ need to be able to handle such situations internally and stream
+ topology requests should therefore always be passed on.
+
+ In the case of chan_pjsip for example, it queues a session refresh
+ (re-INVITE) if it is currently in the middle of a transaction or has
+ pending requests (among other reasons).
+
+ Now, ast_channel_request_stream_topology_change always indicates a
+ stream topology request change even if the requested topology equals the
+ channel's topology.
+
+ For res_pjsip_session.c:
+
+ The function resolve_refresh_media_states does not process stream state
+ changes if the delayed active state differs from the current active
+ state. I.e. if the currently active stream state has changed between the
+ time the sip session refresh request was queued and the time it is being
+ processed, the session refresh is ignored. However, res_pjsip_session
+ contains logic that ensures that session refreshes are queued and
+ re-queued correctly if a session refresh is currently not possible. So
+ this check is not necessary and led to some session refreshes being
+ lost.
+
+ Now, a session refresh is done even if the delayed active state differs
+ from the current active state and it is checked whether the delayed
+ pending state differs from the current active - because that means a
+ refresh is necessary.
+
+ Further, the unit test of resolve_refresh_media_states was adapted to
+ reflect the new behavior. I.e. the changes to delayed pending are
+ prioritized over the changes to current active because we want to
+ preserve the original intention of the pending state.
+
+ ASTERISK-30184
+
+
+#### sla: Prevent deadlock and crash due to autoservicing.
+ Author: Naveen Albert
+ Date: 2022-09-24
+
+ SLAStation currently autoservices the station channel before
+ creating a thread to actually dial the trunk. This leads
+ to duplicate servicing of the channel which causes assertions,
+ deadlocks, crashes, and moreover not the correct behavior.
+
+ Removing the autoservice prevents the crash, but if the station
+ hangs up before the trunk answers, the call hangs since the hangup
+ was never serviced on the channel.
+
+ This is fixed by not autoservicing the channel, but instead
+ servicing it in the thread dialing the trunk, since it is doing
+ so synchronously to begin with. Instead of sleeping for 100ms
+ in a loop, we simply use the channel for timing, and abort
+ if it disappears.
+
+ The same issue also occurs with SLATrunk when a call is answered,
+ because ast_answer invokes ast_waitfor_nandfds. Thus, we use
+ ast_raw_answer instead which does not cause any conflict and allows
+ the call to be answered normally without thread blocking issues.
+
+ ASTERISK-29998 #close
+
+
+#### Build system: Avoid executable stack.
+ Author: Jaco Kroon
+ Date: 2022-11-07
+
+ Found in res_geolocation, but I believe others may have similar issues,
+ thus not linking to a specific issue.
+
+ Essentially gcc doesn't mark the stack for being non-executable unless
+ it's compiling the source, this informs ld via gcc to mark the object as
+ not requiring an executable stack (which a binary blob obviously
+ doesn't).
+
+ ASTERISK-30321
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### func_json: Fix memory leak.
+ Author: Naveen Albert
+ Date: 2022-11-10
+
+ A memory leak was present in func_json due to
+ using ast_json_free, which just calls ast_free,
+ as opposed to recursively freeing the JSON
+ object as needed. This is now fixed to use the
+ right free functions.
+
+ ASTERISK-30293 #close
+
+
+#### test_json: Remove duplicated static function.
+ Author: Naveen Albert
+ Date: 2022-11-10
+
+ Removes the function mkstemp_file and uses
+ ast_file_mkftemp from file.h instead.
+
+ ASTERISK-30295 #close
+
+
+#### res_agi: Respect "transmit_silence" option for "RECORD FILE".
+ Author: Joshua C. Colp
+ Date: 2022-11-16
+
+ The "RECORD FILE" command in res_agi has its own
+ implementation for actually doing the recording. This
+ has resulted in it not actually obeying the option
+ "transmit_silence" when recording.
+
+ This change causes it to now send silence if the
+ option is enabled.
+
+ ASTERISK-30314
+
+
+#### app_mixmonitor: Add option to delete files on exit.
+ Author: Naveen Albert
+ Date: 2022-11-03
+
+ Adds an option that allows MixMonitor to delete
+ its copy of any recording files before exiting.
+
+ This can be handy in conjunction with options
+ like m, which copy the file elsewhere, and the
+ original files may no longer be needed.
+
+ ASTERISK-30284 #close
+
+
+#### manager: Update ModuleCheck documentation.
+ Author: Naveen Albert
+ Date: 2022-11-03
+
+ The ModuleCheck XML documentation falsely
+ claims that the module's version number is returned.
+ This has not been the case since 14, since the version
+ number is not available anymore, but the documentation
+ was not changed at the time. It is now updated to
+ reflect this.
+
+ ASTERISK-30285 #close
+
+
+#### file.c: Don't emit warnings on winks.
+ Author: Naveen Albert
+ Date: 2022-11-06
+
+ Adds an ignore case for wink since it should
+ pass through with no warning.
+
+ ASTERISK-30290 #close
+
+
+#### runUnittests.sh: Save coredumps to proper directory
+ Author: George Joseph
+ Date: 2022-11-02
+
+ Fixed the specification of "outputdir" when calling ast_coredumper
+ so the txt files are saved in the correct place.
+
+ ASTERISK-30282
+
+
+#### app_stack: Print proper exit location for PBXless channels.
+ Author: Naveen Albert
+ Date: 2022-10-01
+
+ When gosub is executed on channels without a PBX, the context,
+ extension, and priority are initialized to the channel driver's
+ default location for that endpoint. As a result, the last Return
+ will restore this location and the Gosub logs will print out bogus
+ information about our exit point.
+
+ To fix this, on channels that don't have a PBX, the execution
+ location is left intact on the last return if there are no
+ further stack frames left. This allows the correct location
+ to be printed out to the user, rather than the bogus default
+ context.
+
+ ASTERISK-30076 #close
+
+
+#### chan_rtp: Make usage of ast_rtp_instance_get_local_address clearer
+ Author: George Joseph
+ Date: 2022-11-02
+
+ unicast_rtp_request() was setting the channel variables like this:
+
+ pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
+ ast_sockaddr_stringify_addr(&local_address));
+ ast_rtp_instance_get_local_address(instance, &local_address);
+ pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
+ ast_sockaddr_stringify_port(&local_address));
+
+ ...which made it appear that UNICASTRTP_LOCAL_ADDRESS was being
+ set before local_address was set. In fact, the address part of
+ local_address was set earlier in the function, just not the port.
+ This was confusing however so ast_rtp_instance_get_local_address()
+ is now being called before setting UNICASTRTP_LOCAL_ADDRESS.
+
+ ASTERISK-30281
+
+
+#### res_pjsip: prevent crash on websocket disconnect
+ Author: Mike Bradeen
+ Date: 2022-10-13
+
+ When a websocket (or potentially any stateful connection) is quickly
+ created then destroyed, it is possible that the qualify thread will
+ destroy the transaction before the initialzing thread is finished
+ with it.
+
+ Depending on the timing, this can cause an assertion within pjsip.
+
+ To prevent this, ast_send_stateful_response will now create the group
+ lock and add a reference to it before creating the transaction.
+
+ While this should resolve the crash, there is still the potential that
+ the contact will not be cleaned up properly, see:ASTERISK~29286. As a
+ result, the contact has to 'time out' before it will be removed.
+
+ ASTERISK-28689
+
+
+#### tcptls: Prevent crash when freeing OpenSSL errors.
+ Author: Naveen Albert
+ Date: 2022-10-27
+
+ write_openssl_error_to_log has been erroneously
+ using ast_free instead of free, which will
+ cause a crash when MALLOC_DEBUG is enabled since
+ the memory was not allocated by Asterisk's memory
+ manager. This changes it to use the actual free
+ function directly to avoid this.
+
+ ASTERISK-30278 #close
+
+
+#### res_pjsip_outbound_registration: Allow to use multiple proxies for registration
+ Author: Igor Goncharovsky
+ Date: 2022-09-09
+
+ Current registration code use pjsip_parse_uri to verify outbound_proxy
+ that is different from the reading this option for the endpoint. This
+ made value with multiple proxies invalid for registration pjsip settings.
+ Removing URI validation helps to use registration through multiple proxies.
+
+ ASTERISK-30217 #close
+
+
+#### tests: Fix compilation errors on 32-bit.
+ Author: Naveen Albert
+ Date: 2022-10-23
+
+ Fix compilation errors caused by using size_t
+ instead of uintmax_t and non-portable format
+ specifiers.
+
+ ASTERISK-30273 #close
+
+
+#### res_pjsip: return all codecs on a re-INVITE without SDP
+ Author: Henning Westerholt
+ Date: 2022-08-26
+
+ Currently chan_pjsip on receiving a re-INVITE without SDP will only
+ return the codecs that are previously negotiated and not offering
+ all enabled codecs.
+
+ This causes interoperability issues with different equipment (e.g.
+ from Cisco) for some of our customers and probably also in other
+ scenarios involving 3PCC infrastructure.
+
+ According to RFC 3261, section 14.2 we SHOULD return all codecs
+ on a re-INVITE without SDP
+
+ The PR proposes a new parameter to configure this behaviour:
+ all_codecs_on_empty_reinvite. It includes the code, documentation,
+ alembic migrations, CHANGES file and example configuration additions.
+
+ ASTERISK-30193 #close
+
+
+#### res_pjsip_notify: Add option support for AMI.
+ Author: Naveen Albert
+ Date: 2022-10-14
+
+ The PJSIP notify CLI commands allow for using
+ "options" configured in pjsip_notify.conf.
+
+ This allows these same options to be used in
+ AMI actions as well.
+
+ Additionally, as part of this improvement,
+ some repetitive common code is refactored.
+
+ ASTERISK-30263 #close
+
+
+#### res_pjsip_logger: Add method-based logging option.
+ Author: Naveen Albert
+ Date: 2022-07-21
+
+ Expands the pjsip logger to support the ability to filter
+ by SIP message method. This can make certain types of SIP debugging
+ easier by only logging messages of particular method(s).
+
+ ASTERISK-30146 #close
+
+ Co-authored-by: Sean Bright <sean@seanbright.com>
+
+#### Dialing API: Cancel a running async thread, may not cancel all calls
+ Author: Frederic LE FOLL
+ Date: 2022-10-06
+
+ race condition: ast_dial_join() may not cancel outgoing call, if
+ function is called just after called party answer and before
+ application execution (bit is_running_app not yet set).
+
+ This fix adds ast_softhangup() calls in addition to existing
+ pthread_kill() when is_running_app is not set.
+
+ ASTERISK-30258
+
+
+#### chan_dahdi: Fix unavailable channels returning busy.
+ Author: Naveen Albert
+ Date: 2022-10-23
+
+ This fixes dahdi_request to properly set the cause
+ code to CONGESTION instead of BUSY if no channels
+ were actually available.
+
+ Currently, the cause is erroneously set to busy
+ if the channel itself is found, regardless of its
+ current state. However, if the channel is not available
+ (e.g. T1 down, card not operable, etc.), then the
+ channel itself may not be in a functional state,
+ in which case CHANUNAVAIL is the correct cause to use.
+
+ This adds a simple check to ensure that busy tone
+ is only returned if a channel is encountered that
+ has an owner, since that is the only possible way
+ that a channel could actually be busy.
+
+ ASTERISK-30274 #close
+
+
+#### res_pjsip_pubsub: Prevent removing subscriptions.
+ Author: Naveen Albert
+ Date: 2022-10-16
+
+ pjproject does not provide any mechanism of removing
+ event packages, which means that once a subscription
+ handler is registered, it is effectively permanent.
+
+ pjproject will assert if the same event package is
+ ever registered again, so currently unloading and
+ loading any Asterisk modules that use subscriptions
+ will cause a crash that is beyond our control.
+
+ For that reason, we now prevent users from being
+ able to unload these modules, to prevent them
+ from ever being loaded twice.
+
+ ASTERISK-30264 #close
+
+
+#### say: Don't prepend ampersand erroneously.
+ Author: Naveen Albert
+ Date: 2022-09-28
+
+ Some logic in say.c for determining if we need
+ to also add an ampersand for file seperation was faulty,
+ as non-successful files would increment the count, causing
+ a leading ampersand to be added improperly.
+
+ This is fixed, and a unit test that captures this regression
+ is also added.
+
+ ASTERISK-30248 #close
+
+
+#### res_crypto: handle unsafe private key files
+ Author: Philip Prindeville
+ Date: 2022-09-16
+
+ ASTERISK-30213 #close
+
+
+#### audiohook: add directional awareness
+ Author: Mike Bradeen
+ Date: 2022-09-29
+
+ Add enum to allow setting optional direction. If set to only one
+ direction, only feed matching-direction frames to the associated
+ slin factory.
+
+ This prevents mangling the transcoder on non-mixed frames when the
+ READ and WRITE frames would have otherwise required it. Also
+ removes the need to mute or discard the un-wanted frames as they
+ are no longer added in the first place.
+
+ res_stasis_snoop is changed to use this addition to set direction
+ on audiohook based on spy direction.
+
+ If no direction is set, the ast_audiohook_init will init this enum
+ to BOTH which maintains existing functionality.
+
+ ASTERISK-30252
+
+
+#### cdr: Allow bridging and dial state changes to be ignored.
+ Author: Naveen Albert
+ Date: 2022-06-01
+
+ Allows bridging, parking, and dial messages to be globally
+ ignored for all CDRs such that only a single CDR record
+ is generated per channel.
+
+ This is useful when CDRs should endure for the lifetime of
+ an entire channel and bridging and dial updates in the
+ dialplan should not result in multiple CDR records being
+ created for the call. With the ignore bridging option,
+ bridging changes have no impact on the channel's CDRs.
+ With the ignore dial state option, multiple Dials and their
+ outcomes have no impact on the channel's CDRs. The
+ last disposition on the channel is preserved in the CDR,
+ so the actual disposition of the call remains available.
+
+ These two options can reduce the amount of "CDR hacks" that
+ have hitherto been necessary to ensure that CDR was not
+ "spoiled" by these messages if that was undesired, such as
+ putting a dummy optimization-disabled local channel between
+ the caller and the actual call and putting the CDR on the channel
+ in the middle to ensure that CDR would persist for the entire
+ call and properly record start, answer, and end times.
+ Enabling these options is desirable when calls correspond
+ to the entire lifetime of channels and the CDR should
+ reflect that.
+
+ Current default behavior remains unchanged.
+
+ ASTERISK-30091 #close
+
+
+#### res_tonedetect: Add ringback support to TONE_DETECT.
+ Author: Naveen Albert
+ Date: 2022-09-30
+
+ Adds support for detecting audible ringback tone
+ to the TONE_DETECT function using the p option.
+
+ ASTERISK-30254 #close
+
+
+#### chan_dahdi: Resolve format truncation warning.
+ Author: Naveen Albert
+ Date: 2022-10-01
+
+ Fixes a format truncation warning in notify_message.
+
+ ASTERISK-30256 #close
+
+
+#### res_crypto: don't modify fname in try_load_key()
+ Author: Philip Prindeville
+ Date: 2022-09-16
+
+ "fname" is passed in as a const char *, but strstr() mangles that
+ into a char *, and we were attempting to modify the string in place.
+ This is an unwanted (and undocumented) side-effect.
+
+ ASTERISK-30213
+
+
+#### res_crypto: use ast_file_read_dirs() to iterate
+ Author: Philip Prindeville
+ Date: 2022-09-15
+
+ ASTERISK-30213
+
+
+#### res_geolocation: Update wiki documentation
+ Author: George Joseph
+ Date: 2022-09-27
+
+ Also added a note to the geolocation.conf.sample file
+ and added a README to the res/res_geolocation/wiki
+ directory.
+
+
+#### res_pjsip: Add mediasec capabilities.
+ Author: Maximilian Fridrich
+ Date: 2022-07-26
+
+ This patch adds support for mediasec SIP headers and SDP attributes.
+ These are defined in RFC 3329, 3GPP TS 24.229 and
+ draft-dawes-sipcore-mediasec-parameter. The new features are
+ implemented so that a backbone for RFC 3329 is present to streamline
+ future work on RFC 3329.
+
+ With this patch, Asterisk can communicate with Deutsche Telekom trunks
+ which require these fields.
+
+ ASTERISK-30032
+
+
+#### res_prometheus: Do not crash on invisible bridges
+ Author: Holger Hans Peter Freyther
+ Date: 2022-09-20
+
+ Avoid crashing by skipping invisible bridges and checking the
+ snapshot for a null pointer. In effect this is how the bridges
+ are enumerated in res/ari/resource_bridges.c already.
+
+ ASTERISK-30239
+ ASTERISK-30237
+
+
+#### res_pjsip_geolocation: Change some notices to debugs.
+ Author: Naveen Albert
+ Date: 2022-09-19
+
+ If geolocation is not in use for an endpoint, the NOTICE
+ log level is currently spammed with messages about this,
+ even though nothing is wrong and these messages provide
+ no real value. These log messages are therefore changed
+ to debugs.
+
+ ASTERISK-30241 #close
+
+
+#### db: Fix incorrect DB tree count for AMI.
+ Author: Naveen Albert
+ Date: 2022-09-24
+
+ The DBGetTree AMI action's ListItem previously
+ always reported 1, regardless of the count. This
+ is corrected to report the actual count.
+
+ ASTERISK-30245 #close
+ patches:
+ gettreecount.diff submitted by Birger Harzenetter (license 5870)
+
+
+#### func_logic: Don't emit warning if both IF branches are empty.
+ Author: Naveen Albert
+ Date: 2022-09-21
+
+ The IF function currently emits warnings if both IF branches
+ are empty. However, there is no actual necessity that either
+ branch be non-empty as, unlike other conditional applications/
+ functions, nothing is inherently done with IF, and both
+ sides could legitimately be empty. The warning is thus turned
+ into a debug message.
+
+ ASTERISK-30243 #close
+
+
+#### features: Add no answer option to Bridge.
+ Author: Naveen Albert
+ Date: 2022-09-11
+
+ Adds the n "no answer" option to the Bridge application
+ so that answer supervision can not automatically
+ be provided when Bridge is executed.
+
+ Additionally, a mechanism (dialplan variable)
+ is added to prevent bridge targets (typically the
+ target of a masquerade) from answering the channel
+ when they enter the bridge.
+
+ ASTERISK-30223 #close
+
+
+#### app_bridgewait: Add option to not answer channel.
+ Author: Naveen Albert
+ Date: 2022-09-09
+
+ Adds the n option to not answer the channel when calling
+ BridgeWait, so the application can be used without
+ forcing answer supervision.
+
+ ASTERISK-30216 #close
+
+
+#### app_amd: Add option to play audio during AMD.
+ Author: Naveen Albert
+ Date: 2022-08-15
+
+ Adds an option that will play an audio file
+ to the party while AMD is running on the
+ channel, so the called party does not just
+ hear silence.
+
+ ASTERISK-30179 #close
+
+
+#### test: initialize capture structure before freeing
+ Author: Philip Prindeville
+ Date: 2022-09-15
+
+ ASTERISK-30232 #close
+
+
+#### func_export: Add EXPORT function
+ Author: Naveen Albert
+ Date: 2021-05-17
+
+ Adds the EXPORT function, which allows write
+ access to variables and functions on other
+ channels.
+
+ ASTERISK-29432 #close
+
+
+#### res_pjsip: Add 100rel option "peer_supported".
+ Author: Maximilian Fridrich
+ Date: 2022-07-26
+
+ This patch adds a new option to the 100rel parameter for pjsip
+ endpoints called "peer_supported". When an endpoint with this option
+ receives an incoming request and the request indicated support for the
+ 100rel extension, then Asterisk will send 1xx responses reliably. If
+ the request did not indicate 100rel support, Asterisk sends 1xx
+ responses normally.
+
+ ASTERISK-30158
+
+
+#### func_scramble: Fix null pointer dereference.
+ Author: Naveen Albert
+ Date: 2022-09-10
+
+ Fix segfault due to null pointer dereference
+ inside the audiohook callback.
+
+ ASTERISK-30220 #close
+
+
+#### manager: be more aggressive about purging http sessions.
+ Author: Jaco Kroon
+ Date: 2022-09-05
+
+ If we find that n_max (currently hard wired to 1) sessions were purged,
+ schedule the next purge for 1ms into the future rather than 5000ms (as
+ per current). This way we will purge up to 1000 sessions per second
+ rather than 1 every 5 seconds.
+
+ This mitigates a build-up of sessions should http sessions gets
+ established faster than 1 per 5 seconds.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### func_strings: Add trim functions.
+ Author: Naveen Albert
+ Date: 2022-09-11
+
+ Adds TRIM, LTRIM, and RTRIM, which can be used
+ for trimming leading and trailing whitespace
+ from strings.
+
+ ASTERISK-30222 #close
+
+
+#### res_crypto: Memory issues and uninitialized variable errors
+ Author: George Joseph
+ Date: 2022-09-16
+
+ ASTERISK-30235
+
+
+#### res_geolocation: Fix issues exposed by compiling with -O2
+ Author: George Joseph
+ Date: 2022-09-16
+
+ Fixed "may be used uninitialized" errors in geoloc_config.c.
+
+ ASTERISK-30234
+
+
+#### res_crypto: don't complain about directories
+ Author: Philip Prindeville
+ Date: 2022-09-13
+
+ ASTERISK-30226 #close
+
+
+#### res_pjsip: Add user=phone on From and PAID for usereqphone=yes
+ Author: Mike Bradeen
+ Date: 2022-08-15
+
+ Adding user=phone to local-side uri's when user_eq_phone=yes is set for
+ an endpoint. Previously this would only add the header to the To and R-URI.
+
+ ASTERISK-30178
+
+
+#### res_geolocation: Fix segfault when there's an empty element
+ Author: George Joseph
+ Date: 2022-09-13
+
+ Fixed a segfault caused by var_list_from_loc_info() encountering
+ an empty location info element.
+
+ Fixed an issue in ast_strsep() where a value with only whitespace
+ wasn't being preserved.
+
+ Fixed an issue in ast_variable_list_from_quoted_string() where
+ an empty value was considered a failure.
+
+ ASTERISK-30215
+ Reported by: Dan Cropp
+
+
+#### res_musiconhold: Add option to not play music on hold on unanswered channels
+ Author: sungtae kim
+ Date: 2022-08-14
+
+ This change adds an option, answeredonly, that will prevent music on
+ hold on channels that are not answered.
+
+ ASTERISK-30135
+
+
+#### res_pjsip: Add TEL URI support for basic calls.
+ Author: Ben Ford
+ Date: 2022-08-02
+
+ This change allows TEL URI requests to come through for basic calls. The
+ allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To
+ headers will now allow TEL URIs, as well as the request URI.
+
+ Support is only for TEL URIs present in traffic from a remote party.
+ Asterisk does not generate any TEL URIs on its own.
+
+ ASTERISK-26894
+
+
+#### res_crypto: Use EVP API's instead of legacy API's
+ Author: Philip Prindeville
+ Date: 2022-03-24
+
+ ASTERISK-30046 #close
+
+
+#### test: Add coverage for res_crypto
+ Author: Philip Prindeville
+ Date: 2022-05-03
+
+ We're validating the following functionality:
+
+ encrypting a block of data with RSA
+ decrypting a block of data with RSA
+ signing a block of data with RSA
+ verifying a signature with RSA
+ encrypting a block of data with AES-ECB
+ encrypting a block of data with AES-ECB
+
+ as well as accessing test keys from the keystore.
+
+ ASTERISK-30045 #close
+
+
+#### res_crypto: make keys reloadable on demand for testing
+ Author: Philip Prindeville
+ Date: 2022-07-26
+
+ ASTERISK-30045
+
+
+#### test: Add test coverage for capture child process output
+ Author: Philip Prindeville
+ Date: 2022-05-03
+
+ ASTERISK-30037 #close
+
+
+#### main/utils: allow checking for command in $PATH
+ Author: Philip Prindeville
+ Date: 2022-07-26
+
+ ASTERISK-30037
+
+
+#### test: Add ability to capture child process output
+ Author: Philip Prindeville
+ Date: 2022-05-02
+
+ ASTERISK-30037
+
+
+#### res_crypto: Don't load non-regular files in keys directory
+ Author: Philip Prindeville
+ Date: 2022-04-26
+
+ ASTERISK-30046
+
+
+#### func_frame_trace: Remove bogus assertion.
+ Author: Naveen Albert
+ Date: 2022-09-08
+
+ The FRAME_TRACE function currently asserts if it sees
+ a MASQUERADE_NOTIFY. However, this is a legitimate thing
+ that can happen so asserting is inappropriate, as there
+ are no clear negative ramifications of such a thing. This
+ is adjusted to be like the other frames to print out
+ the subclass.
+
+ ASTERISK-30210 #close
+
+
+#### lock.c: Add AMI event for deadlocks.
+ Author: Naveen Albert
+ Date: 2022-07-27
+
+ Adds an AMI event to indicate that a deadlock
+ has likely started, when Asterisk is compiled
+ with DETECT_DEADLOCKS enabled. This can make
+ it easier to perform automated deadlock detection
+ and take appropriate action (such as doing a core
+ dump). Unlike the deadlock warnings, the AMI event
+ is emitted only once per deadlock.
+
+ ASTERISK-30161 #close
+
+
+#### app_confbridge: Add end_marked_any option.
+ Author: Naveen Albert
+ Date: 2022-09-04
+
+ Adds the end_marked_any option, which can be used
+ to kick a user from a conference if any marked user
+ leaves.
+
+ ASTERISK-30211 #close
+
+
+#### pbx_variables: Use const char if possible.
+ Author: Naveen Albert
+ Date: 2022-09-03
+
+ Use const char for char arguments to
+ pbx_substitute_variables_helper_full_location
+ that can do so (context and exten).
+
+ ASTERISK-30209 #close
+
+
+#### res_geolocation: Add two new options to GEOLOC_PROFILE
+ Author: George Joseph
+ Date: 2022-08-25
+
+ Added an 'a' option to the GEOLOC_PROFILE function to allow
+ variable lists like location_info_refinement to be appended
+ to instead of replacing the entire list.
+
+ Added an 'r' option to the GEOLOC_PROFILE function to resolve all
+ variables before a read operation and after a Set operation.
+
+ Added a few missing parameters to the ones allowed for writing
+ with GEOLOC_PROFILE.
+
+ Fixed a bug where calling GEOLOC_PROFILE to read a parameter
+ might actually update the profile object.
+
+ Cleaned up XML documentation a bit.
+
+ ASTERISK-30190
+
+
+#### res_geolocation: Allow location parameters on the profile object
+ Author: George Joseph
+ Date: 2022-08-18
+
+ You can now specify the location object's format, location_info,
+ method, location_source and confidence parameters directly on
+ a profile object for simple scenarios where the location
+ information isn't common with any other profiles. This is
+ mutually exclusive with setting location_reference on the
+ profile.
+
+ Updated appdocsxml.dtd to allow xi:include in a configObject
+ element. This makes it easier to link to complete configOptions
+ in another object. This is used to add the above fields to the
+ profile object without having to maintain the option descriptions
+ in two places.
+
+ ASTERISK-30185
+
+
+#### res_geolocation: Add profile parameter suppress_empty_ca_elements
+ Author: George Joseph
+ Date: 2022-08-17
+
+ Added profile parameter "suppress_empty_ca_elements" that
+ will cause Civic Address elements that are empty to be
+ suppressed from the outgoing PIDF-LO document.
+
+ Fixed a possible SEGV if a sub-parameter value didn't have a
+ value.
+
+ ASTERISK-30177
+
+
+#### res_geolocation: Add built-in profiles
+ Author: George Joseph
+ Date: 2022-08-16
+
+ The trigger to perform outgoing geolocation processing is the
+ presence of a geoloc_outgoing_call_profile on an endpoint. This
+ is intentional so as to not leak location information to
+ destinations that shouldn't receive it. In a totally dynamic
+ configuration scenario however, there may not be any profiles
+ defined in geolocation.conf. This makes it impossible to do
+ outgoing processing without defining a "dummy" profile in the
+ config file.
+
+ This commit adds 4 built-in profiles:
+ "<prefer_config>"
+ "<discard_config>"
+ "<prefer_incoming>"
+ "<discard_incoming>"
+ The profiles are empty except for having their precedence
+ set and can be set on an endpoint to allow processing without
+ entries in geolocation.conf. "<discard_config>" is actually the
+ best one to use in this situation.
+
+ ASTERISK-30182
+
+
+#### res_pjsip_sdp_rtp: Skip formats without SDP details.
+ Author: Joshua C. Colp
+ Date: 2022-08-30
+
+ When producing an outgoing SDP we iterate through the configured
+ formats and produce SDP information. It is possible for some
+ configured formats to not have SDP information available. If this
+ is the case we skip over them to allow the SDP to still be
+ produced.
+
+ ASTERISK-29185
+
+
+#### cli: Prevent assertions on startup from bad ao2 refs.
+ Author: Naveen Albert
+ Date: 2022-05-03
+
+ If "core show channels" is run before startup has completed, it
+ is possible for bad ao2 refs to occur because the system is not
+ yet fully initialized. This will lead to an assertion failing.
+
+ To prevent this, initialization of CLI builtins is moved to be
+ later along in the main load sequence. Core CLI commands are
+ loaded at the same time, but channel-related commands are loaded
+ later on.
+
+ ASTERISK-29846 #close
+
+
+#### pjsip: Add TLS transport reload support for certificate and key.
+ Author: Joshua C. Colp
+ Date: 2022-08-19
+
+ This change adds support using the pjsip_tls_transport_restart
+ function for reloading the TLS certificate and key, if the filenames
+ remain unchanged. This is useful for Let's Encrypt and other
+ situations. Note that no restart of the transport will occur if
+ the certificate and key remain unchanged.
+
+ ASTERISK-30186
+
+
+#### res_tonedetect: Fix typos referring to wrong variables.
+ Author: Naveen Albert
+ Date: 2022-08-25
+
+ Fixes two typos that cause fax detection to not work.
+ One refers to the wrong frame variable, and the other
+ refers to the subclass.integer instead of the frametype
+ as it should.
+
+ ASTERISK-30192 #close
+
+
+#### alembic: add missing ps_endpoints columns
+ Author: Mike Bradeen
+ Date: 2022-08-17
+
+ The following required columns were missing,
+ now added to the ps_endpoints table:
+
+ incoming_call_offer_pref
+ outgoing_call_offer_pref
+ stir_shaken_profile
+
+ ASTERISK-29453
+
+
+#### chan_dahdi.c: Resolve a format-truncation build warning.
+ Author: Sean Bright
+ Date: 2022-08-19
+
+ With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0:
+
+ > chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated
+ > writing up to 255 bytes into a region of size between 242 and 252
+ > [-Werror=format-truncation=]
+
+ This removes the error-prone sizeof(...) calculations in favor of just
+ doubling the size of the base buffer.
+
+
+#### res_pjsip_pubsub: Postpone destruction of old subscriptions on RLS update
+ Author: Alexei Gradinari
+ Date: 2022-08-03
+
+ Set termination state to old subscriptions to prevent queueing and sending
+ NOTIFY messages on exten/device state changes.
+
+ Postpone destruction of old subscriptions until all already queued tasks
+ that may be using old subscriptions have completed.
+
+ ASTERISK-29906
+
+
+#### channel.h: Remove redundant declaration.
+ Author: Sean Bright
+ Date: 2022-08-15
+
+ The DECLARE_STRINGFIELD_SETTERS_FOR() declares ast_channel_name_set()
+ for us, so no need to declare it separately.
+
+
+#### features: Add transfer initiation options.
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ Adds additional control options over the transfer
+ feature functionality to give users more control
+ in how the transfer feature sounds and works.
+
+ First, the "transfer" sound that plays when a transfer is
+ initiated can now be customized by the user in
+ features.conf, just as with the other transfer sounds.
+
+ Secondly, the user can now specify the transfer extension
+ in advance by using the TRANSFER_EXTEN variable. If
+ a valid extension is contained in this variable, the call
+ will automatically be transferred to this destination.
+ Otherwise, it will fall back to collecting the extension
+ from the user as is always done now.
+
+ ASTERISK-29899 #close
+
+
+#### CI: Fixing path issue on venv check
+ Author: Mike Bradeen
+ Date: 2022-08-31
+
+ ASTERISK-26826
+
+
+#### CI: use Python3 virtual environment
+ Author: Mike Bradeen
+ Date: 2022-08-11
+
+ Requires Python3 testsuite changes
+
+ ASTERISK-26826
+
+
+#### general: Very minor coding guideline fixes.
+ Author: Naveen Albert
+ Date: 2022-07-28
+
+ Fixes a few coding guideline violations:
+ * Use of C99 comments
+ * Opening brace on same line as function prototype
+
+ ASTERISK-30163 #close
+
+
+#### res_geolocation: Address user issues, remove complexity, plug leaks
+ Author: George Joseph
+ Date: 2022-08-05
+
+ * Added processing for the 'confidence' element.
+ * Added documentation to some APIs.
+ * removed a lot of complex code related to the very-off-nominal
+ case of needing to process multiple location info sources.
+ * Create a new 'ast_geoloc_eprofile_to_pidf' API that just takes
+ one eprofile instead of a datastore of multiples.
+ * Plugged a huge leak in XML processing that arose from
+ insufficient documentation by the libxml/libxslt authors.
+ * Refactored stylesheets to be more efficient.
+ * Renamed 'profile_action' to 'profile_precedence' to better
+ reflect it's purpose.
+ * Added the config option for 'allow_routing_use' which
+ sets the value of the 'Geolocation-Routing' header.
+ * Removed the GeolocProfileCreate and GeolocProfileDelete
+ dialplan apps.
+ * Changed the GEOLOC_PROFILE dialplan function as follows:
+ * Removed the 'profile' argument.
+ * Automatically create a profile if it doesn't exist.
+ * Delete a profile if 'inheritable' is set to no.
+ * Fixed various bugs and leaks
+ * Updated Asterisk WiKi documentation.
+
+ ASTERISK-30167
+
+
+#### chan_iax2: Add missing options documentation.
+ Author: Naveen Albert
+ Date: 2022-07-30
+
+ Adds missing dial resource option documentation.
+
+ ASTERISK-30164 #close
+
+
+#### app_confbridge: Fix memory leak on updated menu options.
+ Author: Naveen Albert
+ Date: 2022-08-01
+
+ If the CONFBRIDGE function is used to dynamically set
+ menu options, a memory leak occurs when a menu option
+ that has been set is overridden, since the menu entry
+ is not destroyed before being freed. This ensures that
+ it is.
+
+ Additionally, logic that duplicates the destroy function
+ is removed in lieu of the destroy function itself.
+
+ ASTERISK-28422 #close
+
+
+#### Geolocation: Wiki Documentation
+ Author: George Joseph
+ Date: 2022-07-19
+
+
+#### manager: Remove documentation for nonexistent action.
+ Author: Naveen Albert
+ Date: 2022-07-28
+
+ The manager XML documentation documents a "FilterList"
+ action, but there is no such action. Therefore, this can
+ lead to confusion when people try to use a documented
+ action that does not, in fact, exist. This is removed
+ as the action never did exist in the past, nor would it
+ be trivial to add since we only store the regex_t
+ objects, so the filter list can't actually be provided
+ without storing that separately. Most likely, the
+ documentation was originally added (around version 10)
+ in anticipation of something that never happened.
+
+ ASTERISK-29917 #close
+
+
+#### general: Improve logging levels of some log messages.
+ Author: Naveen Albert
+ Date: 2022-07-22
+
+ Adjusts some logging levels to be more or less important,
+ that is more prominent when actual problems occur and less
+ prominent for less noteworthy things.
+
+ ASTERISK-30153 #close
+
+
+#### cdr.conf: Remove obsolete app_mysql reference.
+ Author: Naveen Albert
+ Date: 2022-07-27
+
+ The CDR sample config still mentions that app_mysql
+ is available in the addons directory, but this is
+ incorrect as it was removed as of 19. This removes
+ that to avoid confusion.
+
+ ASTERISK-30160 #close
+
+
+#### general: Remove obsolete SVN references.
+ Author: Naveen Albert
+ Date: 2022-07-27
+
+ There are a handful of files in the tree that
+ reference an SVN link for the coding guidelines.
+
+ This removes these because the links are dead
+ and the vast majority of source files do not
+ contain these links, so this is more consistent.
+
+ app_skel still maintains an (up to date) link
+ to the coding guidelines.
+
+ ASTERISK-30159 #close
+
+
+#### app_confbridge: Add missing AMI documentation.
+ Author: Naveen Albert
+ Date: 2022-07-23
+
+ Documents the ConfbridgeListRooms AMI response,
+ which is currently not documented.
+
+ ASTERISK-30020 #close
+
+
+#### app_meetme: Add missing AMI documentation.
+ Author: Naveen Albert
+ Date: 2022-07-23
+
+ The MeetmeList and MeetmeListRooms AMI
+ responses are currently completely undocumented.
+ This adds documentation for these responses.
+
+ ASTERISK-30018 #close
+
+
+#### func_srv: Document field parameter.
+ Author: Naveen Albert
+ Date: 2022-07-23
+
+ Adds missing documentation for the field parameter
+ for the SRVRESULT function.
+
+ ASTERISK-30151
+ Reported by: Chris Young
+
+
+#### pbx_functions.c: Manually update ast_str strlen.
+ Author: Naveen Albert
+ Date: 2022-07-23
+
+ When ast_func_read2 is used to read a function using
+ its read function (as opposed to a native ast_str read2
+ function), the result is copied directly by the function
+ into the ast_str buffer. As a result, the ast_str length
+ remains initialized to 0, which is a bug because this is
+ not the real string length.
+
+ This can cascade and have issues elsewhere, such as when
+ reading substrings of functions that only register read
+ as opposed to read2 callbacks. In this case, since reading
+ ast_str_strlen returns 0, the returned substring is empty
+ as opposed to the actual substring. This has caused
+ the ast_str family of functions to behave inconsistently
+ and erroneously, in contrast to the pbx_variables substitution
+ functions which work correctly.
+
+ This fixes this issue by manually updating the ast_str length
+ when the result is copied directly into the ast_str buffer.
+
+ Additionally, an assertion and a unit test that previously
+ exposed these issues are added, now that the issue is fixed.
+
+ ASTERISK-29966 #close
+
+
+#### build: fix bininstall launchd issue on cross-platform build
+ Author: Sergey V. Lobanov
+ Date: 2022-02-19
+
+ configure script detects /sbin/launchd, but the result of this
+ check is not used in Makefile (bininstall). Makefile also detects
+ /sbin/launchd file to decide if it is required to install
+ safe_asterisk.
+
+ configure script correctly detects cross compile build and sets
+ PBX_LAUNCHD=0
+
+ In case of building asterisk on MacOS host for Linux target using
+ external toolchain (e.g. OpenWrt toolchain), bininstall does not
+ install safe_asterisk (due to /sbin/launchd detection in Makefile),
+ but it is required on target (Linux).
+
+ This patch adds HAVE_SBIN_LAUNCHD=@PBX_LAUNCHD@ to makeopts.in to
+ use the result of /sbin/launchd detection from configure script in
+ Makefile.
+ Also this patch uses HAVE_SBIN_LAUNCHD in Makefile (bininstall) to
+ decide if it is required to install safe_asterisk.
+
+ ASTERISK-29905 #close
+
+
+#### db: Add AMI action to retrieve DB keys at prefix.
+ Author: Naveen Albert
+ Date: 2022-07-11
+
+ Adds the DBGetTree action, which can be used to
+ retrieve all of the DB keys beginning with a
+ particular prefix, similar to the capability
+ provided by the database show CLI command.
+
+ ASTERISK-30136 #close
+
+
+#### manager: Fix incomplete filtering of AMI events.
+ Author: Naveen Albert
+ Date: 2022-07-12
+
+ The global event filtering code was only in one
+ possible execution path, so not all events were
+ being properly filtered out if requested. This moves
+ that into the universal AMI handling code so all
+ events are properly handled.
+
+ Additionally, the CLI listing of disabled events can
+ also get truncated, so we now print out everything.
+
+ ASTERISK-30137 #close
+
+
+#### Update defaultbranch to 20
+ Author: George Joseph
+ Date: 2022-07-20
+
+
+#### res_pjsip: delay contact pruning on Asterisk start
+ Author: Michael Neuhauser
+ Date: 2022-06-14
+
+ Move the call to ast_sip_location_prune_boot_contacts() *after* the call
+ to ast_res_pjsip_init_options_handling() so that
+ res/res_pjsip/pjsip_options.c is informed about the contact deletion and
+ updates its sip_options_contact_statuses list. This allows for an AMI
+ event to be sent by res/res_pjsip/pjsip_options.c if the endpoint
+ registers again from the same remote address and port (i.e., same URI)
+ as used before the Asterisk restart.
+
+ ASTERISK-30109
+ Reported-by: Michael Neuhauser
+
+
+#### chan_dahdi: Fix buggy and missing Caller ID parameters
+ Author: Naveen Albert
+ Date: 2022-03-29
+
+ There are several things wrong with analog Caller ID
+ handling that are fixed by this commit:
+
+ callerid.c's Caller ID generation function contains the
+ logic to use the presentation to properly send the proper
+ Caller ID. However, currently, DAHDI does not pass any
+ presentation information to the Caller ID module, which
+ means that presentation is completely ignored on all calls.
+ This means that lines could be getting Caller ID information
+ they aren't supposed to.
+
+ Part of the reason this has been obscured is because the
+ simple switch logic for handling the built in *67 and *82
+ is completely wrong. Rather than modifying the presentation
+ for the call accordingly (which is what it's supposed to do),
+ it simply blanks out the Caller ID or fills it in. This is
+ wrong, so wrong that it makes a mockery of the specification.
+ Additionally, it would leave to the "UNAVAILABLE" disposition
+ being used for Caller ID generation as opposed to the "PRIVATE"
+ disposition that it should have been using. This is now fixed
+ to only update the presentation and not modify the number and
+ name, so that the simple switch *67/*82 work correctly.
+
+ Next, sig_analog currently only copies over the name and number,
+ nothing else, when it is filling in a duplicated caller id
+ structure. Thus, we also now copy over the presentation
+ information so that is available for the Caller ID spill.
+ Additionally, this meant that "valid" was implicitly 0,
+ and as such presentation would always fail to "Unavailable".
+ The validity is therefore also copied over so it can be used
+ by ast_party_id_presentation.
+
+ As part of this fix, new API is added so that all the relevant
+ Caller ID information can be passed in to the Caller ID generation
+ functions. Parameters that are also completely missing from the
+ Caller ID spill have also been added, to enhance the compatibility,
+ correctness, and completeness of the Asterisk Caller ID implementation.
+
+ ASTERISK-29991 #close
+
+
+#### queues.conf.sample: Correction of typo
+ Author: Sam Banks
+ Date: 2022-07-11
+
+ ASTERISK-30126 #close
+
+
+#### chan_dahdi: Add POLARITY function.
+ Author: Naveen Albert
+ Date: 2022-04-01
+
+ Adds a POLARITY function which can be used to
+ retrieve the current polarity of an FXS channel
+ as well as set the polarity of an FXS channel
+ to idle or reverse at any point during a call.
+
+ ASTERISK-30000 #close
+
+
+#### Makefile: Avoid git-make user conflict
+ Author: Mike Bradeen
+ Date: 2022-06-01
+
+ make_version now silently checks if the required git commands will
+ fail. If they do, then return UNKNOWN__git_check_fail to
+ distinguish this failure from other UNKNOWN__ version failures
+
+ Makefile checks for this value on install and exits out with
+ instructions
+
+ ASTERISK-30029
+
+
+#### app_confbridge: Always set minimum video update interval.
+ Author: Naveen Albert
+ Date: 2022-06-18
+
+ Currently, if multiple video-enabled ConfBridges are
+ conferenced together, we immediately get into a scenario
+ where an infinite sequence of video updates fills up
+ the taskprocessor queue and causes memory consumption
+ to climb unabated until Asterisk is killed. This is due
+ to the core bridging mechanism that provides video updates
+ (softmix_bridge_write_control in bridge_softmix.c)
+ continously updating all the channels in the bridge with
+ video updates.
+
+ The logic to do so in the core is that the video updates
+ should be provided if the video_update_discard property
+ for the bridge is 0, or if enough time has elapsed since
+ the last video update. Thus, we already have a safeguard
+ built in to ensure the scenario described above does not
+ happen. Currently, however, this safeguard is not being
+ adequately ensured.
+
+ In app_confbridge, the video_update_discard property
+ defaults to 2000, which is a healthy value that should
+ completely prevent this issue. However, this value is
+ only set onto the bridge in the SFU video mode. This
+ leaves video modes such as follow_talker completely
+ vulnerable, since video_update_discard will actually
+ be 0, since the default or set value was never applied.
+ As a result, the core bridging mechanism will always
+ try to provide video updates regardless of when the last
+ one was sent.
+
+ To prevent this issue from happening, we now always
+ set the video_update_discard property on the bridge
+ with the value from the bridge profile. The app_confbridge
+ defaults will thus ensure that infinite video updates
+ no longer happen in any video mode.
+
+ ASTERISK-29907 #close
+
+
+#### pbx.c: Simplify ast_context memory management.
+ Author: Sean Bright
+ Date: 2022-07-05
+
+ Allocate all of the ast_context's character data in the structure's
+ flexible array member and eliminate the clunky fake_context. This will
+ simplify future changes to ast_context.
+
+
+#### geoloc_eprofile.c: Fix setting of loc_src in set_loc_src()
+ Author: George Joseph
+ Date: 2022-07-13
+
+ line 196: loc_src = '\0';
+ should have been
+ line 196: *loc_src = '\0';
+
+ The issue was caught by the gcc optimizer complaining that
+ loc_src had a zero length because the pointer itself was being
+ set to NULL instead of the _contents_ of the pointer being set
+ to the NULL terminator.
+
+ ASTERISK-30138
+ Reported-by: Sean Bright
+
+
+#### Geolocation: chan_pjsip Capability Preview
+ Author: George Joseph
+ Date: 2022-07-07
+
+ This commit adds res_pjsip_geolocation which gives chan_pjsip
+ the ability to use the core geolocation capabilities.
+
+ This commit message is intentionally short because this isn't
+ a simple capability. See the documentation at
+ https://wiki.asterisk.org/wiki/display/AST/Geolocation
+ for more information.
+
+ THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
+ USER FEEDBACK!
+
+ ASTERISK-30128
+
+
+#### Geolocation: Core Capability Preview
+ Author: George Joseph
+ Date: 2022-02-15
+
+ This commit adds res_geolocation which creates the core capabilities
+ to manipulate Geolocation information on SIP INVITEs.
+
+ An upcoming commit will add res_pjsip_geolocation which will
+ allow the capabilities to be used with the pjsip channel driver.
+
+ This commit message is intentionally short because this isn't
+ a simple capability. See the documentation at
+ https://wiki.asterisk.org/wiki/display/AST/Geolocation
+ for more information.
+
+ THE CAPABILITIES IMPLEMENTED HERE MAY CHANGE BASED ON
+ USER FEEDBACK!
+
+ ASTERISK-30127
+
+
+#### general: Fix various typos.
+ Author: Naveen Albert
+ Date: 2022-06-01
+
+ ASTERISK-30089 #close
+
+
+#### cel_odbc & res_config_odbc: Add support for SQL_DATETIME field type
+ Author: Kevin Harwell
+ Date: 2022-06-17
+
+ See also: ASTERISK_30023
+
+ ASTERISK-30096 #close
+ patches:
+ inline on issue - submitted by Morvai Szabolcs
+
+
+#### chan_iax2: Allow compiling without OpenSSL.
+ Author: Naveen Albert
+ Date: 2022-07-04
+
+ ASTERISK_30007 accidentally made OpenSSL a
+ required depdendency. This adds an ifdef so
+ the relevant code is compiled only if OpenSSL
+ is available, since it only needs to be executed
+ if OpenSSL is available anyways.
+
+ ASTERISK-30083 #close
+
+
+#### websocket / aeap: Handle poll() interruptions better.
+ Author: Joshua C. Colp
+ Date: 2022-06-28
+
+ A sporadic test failure was happening when executing the AEAP
+ Websocket transport tests. It was originally thought this was
+ due to things not getting cleaned up fast enough, but upon further
+ investigation I determined the underlying cause was poll()
+ getting interrupted and this not being handled in all places.
+
+ This change adds EINTR and EAGAIN handling to the Websocket
+ client connect code as well as the AEAP Websocket transport code.
+ If either occur then the code will just go back to waiting
+ for data.
+
+ The originally disabled failure test case has also been
+ re-enabled.
+
+ ASTERISK-30099
+
+
+#### res_cliexec: Add dialplan exec CLI command.
+ Author: Naveen Albert
+ Date: 2022-05-14
+
+ Adds a CLI command similar to "dialplan eval function" except for
+ applications: "dialplan exec application", useful for quickly
+ testing certain application behavior directly from the CLI
+ without writing any dialplan.
+
+ ASTERISK-30062 #close
+
+
+#### features: Update documentation for automon and automixmon
+ Author: Trevor Peirce
+ Date: 2022-07-03
+
+ The current documentation is out of date and does not reflect actual
+ behaviour. This change makes documentation clearer and accurately
+ reflect the purpose of relevant channel variables.
+
+ ASTERISK-30123
+
+
+#### Geolocation: Base Asterisk Prereqs
+ Author: George Joseph
+ Date: 2022-06-27
+
+ * Added ast_variable_list_from_quoted_string()
+ Parse a quoted string into an ast_variable list.
+
+ * Added ast_str_substitute_variables_full2()
+ Perform variable/function/expression substitution on an ast_str.
+
+ * Added ast_strsep_quoted()
+ Like ast_strsep except you can specify a specific quote character.
+ Also added unit test.
+
+ * Added ast_xml_find_child_element()
+ Find a direct child element by name.
+
+ * Added ast_xml_doc_dump_memory()
+ Dump the specified document to a buffer
+
+ * ast_datastore_free() now checks for a NULL datastore
+ before attempting to destroy it.
+
+
+#### pbx_lua: Remove compiler warnings
+ Author: Boris P. Korzun
+ Date: 2022-06-24
+
+ Improved variable definitions (specified correct type) for avoiding
+ compiler warnings.
+
+ ASTERISK-30117 #close
+
+
+#### res_pjsip_header_funcs: Add functions PJSIP_RESPONSE_HEADER and PJSIP_RESPONSE..
+ Author: Jose Lopes
+ Date: 2022-04-08
+
+ These new functions allow retrieving information from headers on 200 OK
+ INVITE response.
+
+ ASTERISK-29999
+
+
+#### res_prometheus: Optional load res_pjsip_outbound_registration.so
+ Author: Boris P. Korzun
+ Date: 2022-06-09
+
+ Switched res_pjsip_outbound_registration.so dep to optional. Added
+ module loaded check before using it.
+
+ ASTERISK-30101 #close
+
+
+#### app_dial: Fix dial status regression.
+ Author: Naveen Albert
+ Date: 2022-04-30
+
+ ASTERISK_28638 caused a regression by incorrectly aborting
+ early and overwriting the status on certain calls.
+ This was exhibited by certain technologies such as DAHDI,
+ where DAHDI returns NULL for the request if a line is busy.
+ This caused the BUSY condition to be incorrectly treated
+ as CHANUNAVAIL because the DIALSTATUS was getting incorrectly
+ overwritten and call handling was aborted early.
+
+ This is fixed by instead checking if any valid peers have been
+ specified, as opposed to checking the list size of successful
+ requests. This is because the latter could be empty but this
+ does not indicate any kind of problem. This restores the
+ previous working behavior.
+
+ ASTERISK-29989 #close
+
+
+#### db: Notify user if deleted DB entry didn't exist.
+ Author: Naveen Albert
+ Date: 2022-04-01
+
+ Currently, if using the CLI to delete a DB entry,
+ "Database entry removed" is always returned,
+ regardless of whether or not the entry actually
+ existed in the first place. This meant that users
+ were never told if entries did not exist.
+
+ The same issue occurs if trying to delete a DB key
+ using AMI.
+
+ To address this, new API is added that is more stringent
+ in deleting values from AstDB, which will not return
+ success if the value did not exist in the first place,
+ and will print out specific error details if available.
+
+ ASTERISK-30001 #close
+
+
+#### cli: Fix CLI blocking forever on terminating backslash
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ A corner case exists in CLI parsing where if
+ a CLI user in a remote console ends with
+ a backslash and then invokes command completion
+ (using TAB or ?), then the console will freeze
+ forever until a SIGQUIT signal is sent to the
+ process, due to getting blocked forever
+ reading the command completion. CTRL+C
+ and other key combinations have no impact on
+ the CLI session.
+
+ This occurs because, in such cases, the CLI
+ process is waiting for AST_CLI_COMPLETE_EOF
+ to appear in the buffer from the main process,
+ but instead the main process is confused by
+ the funny syntax and thus prints out the CLI help.
+ As a result, the CLI process is stuck on the
+ read call, waiting for the completion that
+ will never come.
+
+ This prevents blocking forever by checking
+ if the data from the main process starts with
+ "Usage:". If it does, that means that CLI help
+ was sent instead of the tab complete vector,
+ and thus the CLI should bail out and not wait
+ any longer.
+
+ ASTERISK-29822 #close
+
+
+#### app_dial: Propagate outbound hook flashes.
+ Author: Naveen Albert
+ Date: 2022-06-18
+
+ The Dial application currently stops hook flashes
+ dead in their tracks from propagating through on
+ outbound calls. This fixes that so they can go
+ down the wire.
+
+ ASTERISK-30115 #close
+
+
+#### res_calendar_icalendar: Send user agent in request.
+ Author: Naveen Albert
+ Date: 2022-06-20
+
+ Microsoft recently began rejecting all requests for
+ ICS calendars on Office 365 with 400 errors if
+ the request doesn't contain a user agent. See:
+
+ https://docs.microsoft.com/en-us/answers/questions/883904/34the-remote-server-returned-an-error-400-bad-requ.html
+
+ Accordingly, we now send a user agent on requests for
+ ICS files so that requests to Office 365 will work as
+ they did before.
+
+ ASTERISK-30106
+
+
+#### say: Abort play loop if caller hangs up.
+ Author: Naveen Albert
+ Date: 2022-05-22
+
+ If the caller has hung up, break out of the play loop so we don't try
+ to play remaining files and fail to do so.
+
+ ASTERISK-30075 #close
+
+
+#### res_pjsip: allow TLS verification of wildcard cert-bearing servers
+ Author: Kevin Harwell
+ Date: 2022-06-08
+
+ Rightly the use of wildcards in certificates is disallowed in accordance
+ with RFC5922. However, RFC2818 does make some allowances with regards to
+ their use when using subject alt names with DNS name types.
+
+ As such this patch creates a new setting for TLS transports called
+ 'allow_wildcard_certs', which when it and 'verify_server' are both enabled
+ allows DNS name types, as well as the common name that start with '*.'
+ to match as a wildcard.
+
+ For instance: *.example.com
+ will match for: foo.example.com
+
+ Partial matching is not allowed, e.g. f*.example.com, foo.*.com, etc...
+ And the starting wildcard only matches for a single level.
+
+ For instance: *.example.com
+ will NOT match for: foo.bar.example.com
+
+ The new setting is disabled by default.
+
+ ASTERISK-30072 #close
+
+
+#### pbx: Add helper function to execute applications.
+ Author: Naveen Albert
+ Date: 2022-05-15
+
+ Finding an application and executing it if found is
+ a common task throughout Asterisk. This adds a helper
+ function around pbx_exec to do this, to eliminate
+ redundant code and make it easier for modules to
+ substitute variables and execute applications by name.
+
+ ASTERISK-30061 #close
+
+
+#### pjsip: Upgrade bundled version to pjproject 2.12.1
+ Author: Stanislav Abramenkov
+ Date: 2022-05-10
+
+ More information:
+ https://github.com/pjsip/pjproject/releases/tag/2.12.1
+
+ Pull request to third-party
+ https://github.com/asterisk/third-party/pull/11
+
+ ASTERISK-30050
+
+
+#### asterisk.c: Fix incompatibility warnings for remote console.
+ Author: Naveen Albert
+ Date: 2022-06-11
+
+ A previous review fixing ASTERISK_22246 and ASTERISK_26582
+ got a couple of the options mixed up as to whether or not
+ they are compatible with the remote console. This fixes
+ those to the best of my knowledge.
+
+ ASTERISK-30097 #close
+
+
+#### test_aeap_transport: disable part of failing unit test
+ Author: Kevin Harwell
+ Date: 2022-06-07
+
+ The 'transport_binary' test sporadically fails, but on a theory that the
+ problem is caused by a previously executed test, transport_connect_fail,
+ part of that test has been disabled until a solution is found.
+
+ ASTERISK_30099
+
+
+#### sig_analog: Fix broken three-way conferencing.
+ Author: Naveen Albert
+ Date: 2022-05-13
+
+ Three-way calling for analog lines is currently broken.
+ If party A is on a call with party B and initiates a
+ three-way call to party C, the behavior differs depending
+ on whether the call is conferenced prior to party C
+ answering. The post-answer case is correct. However,
+ if A flashes before C answers, then the next flash
+ disconnects B rather than C, which is incorrect.
+
+ This error occurs because the subs are not swapped
+ in the misbehaving case. This is because the flash
+ handler only swaps the subs if C has answered already,
+ which is wrong. To fix this, we swap the subs regardless
+ of whether C has answered or not when the call is
+ conferenced. This ensures that C is disconnected
+ on the next hook flash, rather than B as can happen
+ currently.
+
+ ASTERISK-30043 #close
+
+
+#### app_voicemail: Add option to prevent message deletion.
+ Author: Naveen Albert
+ Date: 2022-05-15
+
+ Adds an option to VoiceMailMain that prevents the user
+ from deleting messages during that application invocation.
+ This can be useful for public or shared mailboxes, where
+ some users should be able to listen to messages but not
+ delete them.
+
+ ASTERISK-30063 #close
+
+
+#### res_parking: Add music on hold override option.
+ Author: Naveen Albert
+ Date: 2022-05-31
+
+ An m option to Park and ParkAndAnnounce now allows
+ specifying a music on hold class override.
+
+ ASTERISK-30087
+
+
+#### xmldocs: Improve examples.
+ Author: Naveen Albert
+ Date: 2022-06-01
+
+ Use example tags instead of regular para tags
+ where possible.
+
+ ASTERISK-30090
+
+
+#### res_pjsip_outbound_registration: Make max random delay configurable.
+ Author: Naveen Albert
+ Date: 2022-03-12
+
+ Currently, PJSIP will randomly wait up to 10 seconds for each
+ outbound registration's initial attempt. The reason for this
+ is to avoid having all outbound registrations attempt to register
+ simultaneously.
+
+ This can create limitations with the test suite where we need to
+ be able to receive inbound calls potentially within 10 seconds of
+ starting up. For instance, we might register to another server
+ and then try to receive a call through the registration, but if
+ the registration hasn't happened yet, this will fail, and hence
+ this inconsistent behavior can cause tests to fail. Ultimately,
+ this requires a smaller random value because there may be no good
+ reason to wait for up to 10 seconds in these circumstances.
+
+ To address this, a new config option is introduced which makes this
+ maximum delay configurable. This allows, for instance, this to be
+ set to a very small value in test systems to ensure that registrations
+ happen immediately without an unnecessary delay, and can be used more
+ generally to control how "tight" the initial outbound registrations
+ are.
+
+ ASTERISK-29965 #close
+
+
+#### res_pjsip: Actually enable session timers when timers=always
+ Author: Trevor Peirce
+ Date: 2022-06-07
+
+ When a pjsip endpoint is defined with timers=always, this has been a
+ functional noop. This patch correctly sets the feature bitmap to both
+ enable support for session timers and to enable them even when the
+ endpoint itself does not request or support timers.
+
+ ASTERISK-29603
+ Reported-By: Ray Crumrine
+
+
+#### res_pjsip_pubsub: delete scheduled notification on RLS update
+ Author: Alexei Gradinari
+ Date: 2022-06-06
+
+ If there is scheduled notification, we must delete it
+ to avoid using destroyed subscriptions.
+
+ ASTERISK-29906
+
+
+#### res_pjsip_pubsub: XML sanitized RLS display name
+ Author: Alexei Gradinari
+ Date: 2022-06-07
+
+ ASTERISK-29891
+
+
+#### app_sayunixtime: Use correct inflection for German time.
+ Author: Christof Efkemann
+ Date: 2022-06-01
+
+ In function ast_say_date_with_format_de(), take special
+ care when the hour is one o'clock. In this case, the
+ German number "eins" must be inflected to its neutrum form,
+ "ein". This is achieved by playing "digits/1N" instead of
+ "digits/1". Fixes both 12- and 24-hour formats.
+
+ ASTERISK-30092
+
+
+#### chan_iax2: Prevent deadlock due to duplicate autoservice.
+ Author: Naveen Albert
+ Date: 2022-05-16
+
+ If a switch is invoked using chan_iax2, deadlock can result
+ because the PBX core is autoservicing the channel while chan_iax2
+ also then attempts to service it while waiting for the result
+ of the switch. This removes servicing of the channel to prevent
+ any conflicts.
+
+ ASTERISK-30064 #close
+
+
+#### loader: Prevent deadlock using tab completion.
+ Author: Naveen Albert
+ Date: 2022-05-03
+
+ If tab completion using ast_module_helper is attempted
+ during startup, deadlock will ensue because the CLI
+ will attempt to lock the module list while it is already
+ locked by the loader. This causes deadlock because when
+ the loader tries to acquire the CLI lock, they are blocked
+ on each other.
+
+ Waiting for startup to complete is not feasible because
+ the CLI lock is acquired while waiting, so deadlock will
+ ensure regardless of whether or not a lock on the module
+ list is attempted.
+
+ To prevent deadlock, we immediately abort if tab completion
+ is attempted on the module list before Asterisk is fully
+ booted.
+
+ ASTERISK-30039 #close
+
+
+#### res_calendar: Prevent assertion if event ends in past.
+ Author: Naveen Albert
+ Date: 2022-03-23
+
+ res_calendar will trigger an assertion currently
+ if the ending time is calculated to be in the past.
+ Unlike the reminder and start times, however, there
+ is currently no check to catch non-positive times
+ and set them to 1. As a result, if we get a negative
+ value by happenstance, this can cause a crash.
+
+ To prevent the assertion from begin triggered, we now
+ use the same logic as the reminder and start events
+ to catch this issue before it can cause a problem.
+
+ ASTERISK-29981 #close
+
+
+#### res_parking: Warn if out of bounds parking spot requested.
+ Author: Naveen Albert
+ Date: 2022-05-30
+
+ Emits a warning if the user has requested a parking spot that
+ is out of bounds for the requested parking lot.
+
+ ASTERISK-30086
+
+
+#### chan_pjsip: Only set default audio stream on hold.
+ Author: Maximilian Fridrich
+ Date: 2022-05-19
+
+ When a PJSIP channel is set on hold or off hold, all streams were set
+ on/off hold. This is not the desired behaviour and caused issues
+ when there were multiple streams in the topology.
+
+ Now, only the default audio stream is set on/off hold when a hold is
+ indicated.
+
+ ASTERISK-30051
+
+
+#### res_pjsip_dialog_info_body_generator: Set LOCAL target URI as local URI
+ Author: Alexei Gradinari
+ Date: 2022-05-26
+
+ The change "Add LOCAL/REMOTE tags in dialog-info+xml" set both "local"
+ Identity Element URI and Target Element URI to the same value -
+ the channel Caller Number.
+ For Identity Element it's ok to set as Caller ID.
+ But Local Target URI should be set as local URI.
+
+ In this case the Local Target URI can be used for Directed Call Pickup
+ by Polycom ip-phones (parameter useLocalTargetUriforLegacyPickup).
+
+ Also XML sanitized Display names.
+
+ ASTERISK-24601
+
+
+#### res_agi: Evaluate dialplan functions and variables in agi exec if enabled
+ Author: Shloime Rosenblum
+ Date: 2022-05-11
+
+ Agi commnad exec can now evaluate dialplan functions and
+ variables if variable AGIEXECFULL is set to yes. this can
+ be useful when executing Playback or Read from agi.
+
+ ASTERISK-30058 #close
+
+
+#### ast_pkgconfig.m4: AST_PKG_CONFIG_CHECK() relies on sed.
+ Author: Sean Bright
+ Date: 2022-05-17
+
+ Make sure that we have a working sed before trying to use it.
+
+ ASTERISK-30059 #close
+
+
+#### ari: expose channel driver's unique id to ARI channel resource
+ Author: Moritz Fain
+ Date: 2022-04-26
+
+ This change exposes the channel driver's unique id (i.e. the Call-ID
+ for chan_sip/chan_pjsip based channels) to ARI channel resources
+ as `protocol_id`.
+
+ ASTERISK-30027
+ Reported by: Moritz Fain
+ Tested by: Moritz Fain
+
+
+#### loader.c: Use portable printf conversion specifier for int64.
+ Author: Sean Bright
+ Date: 2022-05-17
+
+ ASTERISK-30060 #close
+
+
+#### res_pjsip_transport_websocket: Also set the remote name.
+ Author: Joshua C. Colp
+ Date: 2022-05-17
+
+ As part of PJSIP 2.11 a behavior change was done to require
+ a matching remote hostname on an established transport for
+ secure transports. Since the Websocket transport is considered
+ a secure transport this caused the existing connection to not
+ be found and used.
+
+ We now set the remote hostname and the transport can be found.
+
+ ASTERISK-30065
+
+
+#### res_pjsip_transport_websocket: save the original contact host
+ Author: Thomas Guebels
+ Date: 2022-05-04
+
+ This is needed to be able to restore it in REGISTER responses,
+ otherwise the client won't be able to find the contact it created.
+
+ ASTERISK-30042
+
+
+#### res_pjsip_outbound_registration: Show time until expiration
+ Author: Naveen Albert
+ Date: 2022-01-07
+
+ Adjusts the pjsip show registration(s) commands to show
+ the amount of seconds remaining until a registration
+ expires.
+
+ ASTERISK-29845 #close
+
+
+#### app_confbridge: Add function to retrieve channels.
+ Author: Naveen Albert
+ Date: 2022-04-29
+
+ Adds the CONFBRIDGE_CHANNELS function which can be used
+ to retrieve a comma-separated list of channels, filtered
+ by a particular type of participant category. This output
+ can then be used with functions like UNSHIFT, SHIFT, POP,
+ etc.
+
+ ASTERISK-30036 #close
+
+
+#### chan_dahdi: Fix broken operator mode clearing.
+ Author: Naveen Albert
+ Date: 2022-04-26
+
+ Currently, the operator services mode in DAHDI is broken and unusable.
+ The actual operator recall functionality works properly; however,
+ when the operator hangs up (which is the only way that such a call
+ is allowed to end), both lines are permanently taken out of service
+ until "dahdi restart" is run. This prevents this feature from being
+ used.
+
+ Operator mode is one of the few factors that can cause the general
+ analog event handling in sig_analog not to be used. Several years
+ back, much of the analog handling was moved from chan_dahdi to
+ sig_analog. However, this was not done fully or consistently at
+ the time, and when operator mode is active, sig_analog does not
+ get used. Generally this is correct, but in the case of hangup
+ it should be using sig_analog regardless of the operator mode;
+ otherwise, the lines do not properly clear and they become unusable.
+
+ This bug is fixed so the operator can now hang up and properly
+ release the call. It is treated just like any other hangup. The
+ operator mode functionality continues to work as it did before.
+
+ ASTERISK-29993 #close
+
+
+#### GCC12: Fixes for 16+
+ Author: George Joseph
+ Date: 2022-05-03
+
+ Most issues were in stringfields and had to do with comparing
+ a pointer to an constant/interned string with NULL. Since the
+ string was a constant, a pointer to it could never be NULL so
+ the comparison was always "true". gcc now complains about that.
+
+ There were also a few issues where determining if there was
+ enough space for a memcpy or s(n)printf which were fixed
+ by defining some of the involved variables as "volatile".
+
+ There were also a few other miscellaneous fixes.
+
+ ASTERISK-30044
+
+
+#### GCC12: Fixes for 18+. state_id_by_topic comparing wrong value
+ Author: George Joseph
+ Date: 2022-05-04
+
+ GCC 12 caught an issue in state_id_by_topic where we were
+ checking a pointer for NULL instead of the contents of
+ the pointer for '\0'.
+
+ ASTERISK-30044
+
+
+#### core_unreal: Flip stream direction of second channel.
+ Author: Maximilian Fridrich
+ Date: 2022-04-29
+
+ When a new unreal (local) channel is created, a second (;2) channel is
+ created as a counterpart which clones the topology of the first
+ channel. This creates issues when an outgoing stream is sendonly or
+ recvonly as the stream state of the inbound channel will be the same
+ as the stream state of the outbound channel.
+
+ Now the stream state is flipped for the streams of the 2nd channel in
+ ast_unreal_new_channels if the outgoing stream topology is recvonly or
+ sendonly.
+
+ ASTERISK-29655
+ Reported by: Michael Auracher
+
+ ASTERISK-29638
+ Reported by: Michael Auracher
+
+
+#### chan_dahdi: Document dial resource options.
+ Author: Naveen Albert
+ Date: 2022-03-27
+
+ Documents the Dial syntax for DAHDI, namely the channel group,
+ distinctive ring, answer confirmation, and digital call options
+ that are specified in the resource itself.
+
+ ASTERISK-24827 #close
+
+
+#### chan_dahdi: Don't allow MWI FSK if channel not idle.
+ Author: Naveen Albert
+ Date: 2022-03-29
+
+ For lines that have mailboxes configured on them, with
+ FSK MWI, DAHDI will periodically try to dispatch FSK
+ to update MWI. However, this is never supposed to be
+ done when a channel is not idle.
+
+ There is currently an edge case where MWI FSK can
+ extraneously get spooled for the channel if a caller
+ hook flashes and hangs up, which triggers a recall ring.
+ After one ring, the on hook time threshold in this if
+ condition has been satisfied and an MWI update is spooled.
+ This means that when the phone is picked up again, the
+ answerer gets an FSK spill before being reconnected to
+ the party on hold.
+
+ To prevent this, we now explicitly check to ensure that
+ subchannel 0 has no owner. There is no owner when DAHDI
+ channels are idle, but if the channel is "in use" in some
+ way (such as in the aforementioned scenario), then there
+ is an owner, and we shouldn't process MWI at this time.
+
+ ASTERISK-28518 #close
+
+
+#### apps/confbridge: Added hear_own_join_sound option to control who hears sound_j..
+ Author: Michael Cargile
+ Date: 2022-02-23
+
+ Added the hear_own_join_sound option to the confbridge user profile to
+ control who hears the sound_join audio file. When set to 'yes' the user
+ entering the conference and the participants already in the conference
+ will hear the sound_join audio file. When set to 'no' the user entering
+ the conference will not hear the sound_join audio file, but the
+ participants already in the conference will hear the sound_join audio
+ file.
+
+ ASTERISK-29931
+ Added by Michael Cargile
+
+
+#### chan_dahdi: Don't append cadences on dahdi restart.
+ Author: Naveen Albert
+ Date: 2022-03-27
+
+ Currently, if any custom ring cadences are specified, they are
+ appended to the array of cadences from wherever we left off
+ last time. This works properly the first time, but on subsequent
+ dahdi restarts, it means that the existing cadences are left
+ alone and (most likely) the same cadences are then re-added
+ afterwards. In short order, the cadence array gets maxed out
+ and the user begins seeing warnings that the array is full
+ and no more cadences may be added.
+
+ This buggy behavior persists until Asterisk is completely
+ restarted; however, if and when dahdi restart is run again,
+ then the same problem is reintroduced.
+
+ This fixes this behavior so that cadence parsing is more
+ idempotent, that is so running dahdi restart multiple times
+ starts adding cadences from the beginning, rather than from
+ wherever the last cadence was added.
+
+ As before, it is still not possible to revert to the default
+ cadences by simply removing all cadences in this manner, nor
+ is it possible to delete existing cadences. However, this
+ does make it possible to update existing cadences, which
+ was not possible before, and also ensures that the cadences
+ remain unchanged if the config remains unchanged.
+
+ ASTERISK-29990 #close
+
+
+#### chan_iax2: Prevent crash if dialing RSA-only call without outkey.
+ Author: Naveen Albert
+ Date: 2022-04-02
+
+ Currently, if attempting to place a call to a peer that only allows
+ RSA authentication, if we fail to provide an outkey when placing
+ the call, Asterisk will crash.
+
+ This exposes the broader issue that IAX2 is prone to causing a crash
+ if encryption or decryption is attempted but we never initialized
+ the encryption and decryption keys. In other words, if the logic
+ to use encryption in chan_iax2 is not perfectly aligned with the
+ decision to build keys in the first place, then a crash is not
+ only possible but probable. This was demonstrated by ASTERISK_29264,
+ for instance.
+
+ This permanently prevents such events from causing a crash by explicitly
+ checking that keys are initialized properly before setting the flags
+ to use encryption for the call. Instead of crashing, the call will
+ now abort.
+
+ ASTERISK-30007 #close
+
+
+#### menuselect: Don't erroneously recompile modules.
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ A bug in menuselect can cause modules that are disabled
+ by default to be recompiled every time a recompilation
+ occurs. This occurs for module categories that are NOT
+ positive output, as for these categories, the modules
+ contained in the makeopts file indicate modules which
+ should NOT be selected. The existing procedure of iterating
+ through these modules to mark modules as present is thus
+ insufficient. This has led to modules with a default_enabled
+ tag of "no" to get deleted and recompiled every time, even
+ when they haven't changed.
+
+ To fix this, we now modify the mark as present behavior
+ for module categories that are not positive output. For
+ these, we start by iterating through the module tree
+ and marking all modules as present, then go back and
+ mark anything contained in the makeopts file as not
+ present. This ensures that makeopt selections are actually
+ used properly, regardless of whether a module category
+ uses positive output or not.
+
+ ASTERISK-29728 #close
+
+
+#### app_meetme: Don't erroneously set global variables.
+ Author: Naveen Albert
+ Date: 2022-03-31
+
+ The admin_exec function in app_meetme is used by the SLA
+ applications for internal bridging. However, in these cases,
+ chan is NULL. Currently, this function will set some status
+ variables that are intended for a channel, but since channel
+ is NULL, this is erroneously creating meaningless global
+ variables, which shouldn't be happening. This sets these
+ variables only if chan is not NULL.
+
+ ASTERISK-30002 #close
+
+
+#### asterisk.c: Warn of incompatibilities with remote console.
+ Author: Naveen Albert
+ Date: 2022-03-05
+
+ Some command line options to Asterisk only apply when Asterisk
+ is started and cannot be used with remote console mode. If a
+ user tries to use any of these, they are currently simply
+ silently ignored.
+
+ This prints out a warning if incompatible options are used,
+ informing users that an option used cannot be used with remote
+ console mode. Additionally, some clarifications are added to
+ the help text and man page.
+
+ ASTERISK-22246
+ ASTERISK-26582
+
+
+#### func_db: Add function to return cardinality at prefix
+ Author: Naveen Albert
+ Date: 2022-03-15
+
+ Adds the DB_KEYCOUNT function, which can be used to retrieve
+ the number of keys at a given prefix in AstDB.
+
+ ASTERISK-29968 #close
+
+
+#### chan_dahdi: Fix insufficient array size for round robin.
+ Author: Naveen Albert
+ Date: 2022-03-30
+
+ According to chan_dahdi.conf, up to 64 groups (numbered
+ 0 through 63) can be used when dialing DAHDI channels.
+
+ However, currently dialing round robin with a group number
+ greater than 31 fails because the array for the round robin
+ structure is only size 32, instead of 64 as it should be.
+
+ This fixes that so the round robin array size is consistent
+ with the actual groups capacity.
+
+ ASTERISK-29994
+
+
+#### chan_sip.c Session timers get removed on UPDATE
+ Author: Mark Petersen
+ Date: 2022-02-26
+
+ If Asterisk receives a SIP REFER with Session-Timers UAC
+ maintain Session-Timers when sending UPDATE"
+
+ ASTERISK-29843
+
+
+#### func_evalexten: Extension evaluation function.
+ Author: Naveen Albert
+ Date: 2021-06-21
+
+ This adds the EVAL_EXTEN function, which may be used to retrieve
+ the variable-substituted data at any extension.
+
+ ASTERISK-29486
+
+
+#### file.c: Prevent formats from seeking negative offsets.
+ Author: Naveen Albert
+ Date: 2022-03-01
+
+ Currently, if a user uses an application like ControlPlayback
+ to try to rewind a file past the beginning, this can throw
+ warnings when the file format (e.g. PCM) tries to seek to
+ a negative offset.
+
+ Instead of letting file formats try (and fail) to seek a
+ negative offset, we instead now catch this in the rewind
+ function to ensure that we never seek an offset less than 0.
+ This prevents legitimate user actions from triggering warnings
+ from any particular file formats.
+
+ ASTERISK-29943 #close
+
+
+#### chan_pjsip: Add ability to send flash events.
+ Author: Naveen Albert
+ Date: 2022-02-26
+
+ PJSIP currently is capable of receiving flash events
+ and converting them to FLASH control frames, but it
+ currently lacks support for doing the reverse: taking
+ a FLASH control frame and converting it into a flash
+ event in the SIP domain.
+
+ This adds the ability for PJSIP to process flash control
+ frames by converting them into the appropriate SIP INFO
+ message, which can then be sent to the peer. This allows,
+ for example, flash events to be sent between Asterisk
+ systems using PJSIP.
+
+ ASTERISK-29941 #close
+
+
+#### cli: Add command to evaluate dialplan functions.
+ Author: Naveen Albert
+ Date: 2021-12-26
+
+ Adds the dialplan eval function commands to evaluate a dialplan
+ function from the CLI. The return value and function result are
+ printed out and can be used for testing or debugging.
+
+ ASTERISK-29820 #close
+
+
+#### documentation: Adds versioning information.
+ Author: Naveen Albert
+ Date: 2022-02-25
+
+ Adds version information for applications, functions,
+ and manager events/actions.
+
+ This is not completely exhaustive by any means but
+ covers most new things added that have release
+ versioning information in the issue tracker.
+
+ ASTERISK-29940 #close
+
+
+#### samples: Remove obsolete sample configs.
+ Author: Naveen Albert
+ Date: 2022-04-02
+
+ Removes a couple sample config files for modules
+ which have since been removed from Asterisk.
+
+ ASTERISK-30008 #close
+
+
+#### chan_pjsip: add allow_sending_180_after_183 option
+ Author: Mark Petersen
+ Date: 2022-02-21
+
+ added new global config option "allow_sending_180_after_183"
+ that if enabled will preserve 180 after a 183
+
+ ASTERISK-29842
+
+
+#### chan_sip: SIP route header is missing on UPDATE
+ Author: Mark Petersen
+ Date: 2022-03-07
+
+ if Asterisk need to send an UPDATE before answer
+ on a channel that uses Record-Route:
+ it will not include a Route header
+
+ ASTERISK-29955
+
+
+#### manager: Terminate session on write error.
+ Author: Joshua C. Colp
+ Date: 2022-04-25
+
+ On a write error to an AMI session a flag was set to
+ indicate that the write error had occurred, with the
+ expected result being that the session be terminated.
+ This was not actually happening and instead writing
+ would continue to be attempted.
+
+ This change adds a check for the write error and causes
+ the session to actually terminate.
+
+ ASTERISK-29948
+
+
+#### bridge_simple.c: Unhold channels on join simple bridge.
+ Author: Yury Kirsanov
+ Date: 2022-04-21
+
+ Patch provided inline by Yury Kirsanov on the linked issue and
+ approved by Josh Colp.
+
+ ASTERISK-29253 #close
+
+
+#### res_aeap & res_speech_aeap: Add Asterisk External Application Protocol
+ Author: Kevin Harwell
+ Date: 2021-06-18
+
+ Add framework to connect to, and read and write protocol based
+ messages from and to an external application using an Asterisk
+ External Application Protocol (AEAP). This has been divided into
+ several abstractions:
+
+ 1. transport - base communication layer (currently websocket only)
+ 2. message - AEAP description and data (currently JSON only)
+ 3. transaction - links/binds requests and responses
+ 4. aeap - transport, message, and transaction handler/manager
+
+ This patch also adds an AEAP implementation for speech to text.
+ Existing speech API callbacks for speech to text have been completed
+ making it possible for Asterisk to connect to a configured external
+ translator service and provide audio for STT. Results can also be
+ received from the external translator, and made available as speech
+ results in Asterisk.
+
+ Unit tests have also been created that test the AEAP framework, and
+ also the speech to text implementation.
+
+ ASTERISK-29726 #close
+
+
+#### app_dial: Flip stream direction of outgoing channel.
+ Author: Maximilian Fridrich
+ Date: 2022-04-13
+
+ When executing dial, the topology of the incoming channel is cloned and
+ used for the outgoing channel. This creates issues when an incoming
+ stream is sendonly or recvonly as the stream state of the outgoing
+ channel will be the same as the stream state of the incoming channel.
+
+ Now the stream state is flipped for the outgoing stream in
+ dial_exec_full if the incoming stream topology is recvonly or sendonly.
+
+ ASTERISK-29655
+ Reported by: Michael Auracher
+
+ ASTERISK-29638
+ Reported by: Michael Auracher
+
+
+#### res_pjsip_stir_shaken.c: Fix enabled when not configured.
+ Author: Ben Ford
+ Date: 2022-04-21
+
+ There was an issue with the conditional where STIR/SHAKEN would be
+ enabled even when not configured. It has been changed to ensure that if
+ a profile does not exist and stir_shaken is not set in pjsip.conf, then
+ the conditional will return from the function without performing
+ STIR/SHAKEN operations.
+
+ ASTERISK-30024
+
+
+#### res_pjsip: Always set async_operations to 1.
+ Author: Joshua C. Colp
+ Date: 2022-04-06
+
+ The async_operations setting on a transport configures how
+ many simultaneous incoming packets the transport can handle
+ when multiple threads are polling and waiting on the transport.
+ As we only use a single thread this was needlessly creating
+ incoming packets when set to a non-default value, wasting memory.
+
+ ASTERISK-30006
+
+
+#### config.h: Don't use C++ keywords as argument names.
+ Author: Sean Bright
+ Date: 2022-04-19
+
+ ASTERISK-30021 #close
+
+
+#### cdr_adaptive_odbc: Add support for SQL_DATETIME field type.
+ Author: Joshua C. Colp
+ Date: 2022-04-20
+
+ ASTERISK-30023
+
+
+#### pjsip: Increase maximum number of format attributes.
+ Author: Joshua C. Colp
+ Date: 2022-04-11
+
+ Chrome has added more attributes, causing the limit to be
+ exceeded. This raises it up some more.
+
+ ASTERISK-30015
+
+
+#### AST-2022-002 - res_stir_shaken/curl: Add ACL checks for Identity header.
+ Author: Ben Ford
+ Date: 2022-02-28
+
+ Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
+ can be specified on a per endpoint basis. This option will reference a
+ stir_shaken_profile that can be configured in stir_shaken.conf. The type
+ of this option must be 'profile'. The stir_shaken option can be
+ specified on this object with the same values as before (attest, verify,
+ on), but it cannot be off since having the profile itself implies wanting
+ STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
+ with permit and deny lines in the object itself) that will be used to
+ limit what interfaces Asterisk will attempt to retrieve information from
+ when reading the Identity header.
+
+ ASTERISK-29476
+
+
+#### AST-2022-001 - res_stir_shaken/curl: Limit file size and check start.
+ Author: Ben Ford
+ Date: 2022-01-07
+
+ Put checks in place to limit how much we will actually download, as well
+ as a check for the data we receive at the start to ensure it begins with
+ what we would expect a certificate to begin with.
+
+ ASTERISK-29872
+
+
+#### func_odbc: Add SQL_ESC_BACKSLASHES dialplan function.
+ Author: Joshua C. Colp
+ Date: 2022-02-10
+
+ Some databases depending on their configuration using backslashes
+ for escaping. When combined with the use of ' this can result in
+ a broken func_odbc query.
+
+ This change adds a SQL_ESC_BACKSLASHES dialplan function which can
+ be used to escape the backslashes.
+
+ This is done as a dialplan function instead of being always done
+ as some databases do not require this, and always doing it would
+ result in incorrect data being put into the database.
+
+ ASTERISK-29838
+
+
+#### app_mf, app_sf: Return -1 if channel hangs up.
+ Author: Naveen Albert
+ Date: 2022-03-05
+
+ The ReceiveMF and ReceiveSF applications currently always
+ return 0, even if a channel has hung up. The call will still
+ end but generally applications are expected to return -1 if
+ the channel has hung up.
+
+ We now return -1 if a hangup occured to bring this behavior
+ in line with this norm. This has no functional impact, but
+ merely increases conformity with how these modules interact
+ with the PBX core.
+
+ ASTERISK-29951 #close
+
+
+#### app_queue: Add music on hold option to Queue.
+ Author: Naveen Albert
+ Date: 2022-01-22
+
+ Adds the m option to the Queue application, which allows a
+ music on hold class to be specified at runtime which will
+ override the class configured in queues.conf.
+
+ This option functions like the m option to Dial.
+
+ ASTERISK-29876 #close
+
+
+#### app_meetme: Emit warning if conference not found.
+ Author: Naveen Albert
+ Date: 2022-03-05
+
+ Currently, if a user tries to access a non-dynamic
+ MeetMe conference and the conference is not found,
+ the call simply silent hangs up. There is no indication
+ to the user that anything went wrong at all.
+
+ This changes the relevant debug message to a warning
+ so that the user is notified of this invalidity.
+
+ ASTERISK-29954 #close
+
+
+#### build: Remove obsolete leftover build references.
+ Author: Naveen Albert
+ Date: 2022-02-24
+
+ Removes some leftover build and config references to
+ modules that have since been removed from Asterisk.
+
+ ASTERISK-29935 #close
+
+
+#### res_pjsip_header_funcs: wrong pool used tdata headers
+ Author: Kevin Harwell
+ Date: 2022-03-23
+
+ When adding headers to an outgoing request the headers were cloned using
+ the dialog's pool when they should have been cloned using tdata's pool.
+ Under certain circumstances it was possible for the dialog object, and
+ its pool to be freed while tdata is still active and available. Thus the
+ cloned header "disappeared", and when tdata tried to later access it a
+ crash would occur.
+
+ This patch makes it so all added headers are cloned appropriately using
+ tdata's pool.
+
+ ASTERISK-29411 #close
+ ASTERISK-29535 #close
+
+
+#### deprecation cleanup: remove leftover files
+ Author: Kevin Harwell
+ Date: 2022-03-25
+
+ Several modules removal and deprecations occurred in 19.0.0 (initial
+ 19 release), but associated UPGRADE files were not removed from
+ staging for some reason in the master branch.
+
+ This patch removes those files, and also removes a spurious leftover
+ header, chan_phone.h (associated module removed in 19).
+
+
+#### pjproject: Update bundled to 2.12 release.
+ Author: Joshua C. Colp
+ Date: 2022-02-24
+
+ This change removes patches which have been merged into
+ upstream and updates some existing ones. It also adds
+ some additional config_site.h changes to restore previous
+ behavior, as well as a patch to allow multiple Authorization
+ headers. There seems to be some confusion or disagreement
+ on language in RFC 8760 in regards to whether multiple
+ Authorization headers are supported. The RFC implies it
+ is allowed, as does some past sipcore discussion. There is
+ also the catch all of "local policy" to allow it. In
+ the case of Asterisk we allow it.
+
+ ASTERISK-29351
+
+
+#### pbx.c: Warn if there are too many includes in a context.
+ Author: Naveen Albert
+ Date: 2022-03-05
+
+ The PBX core uses the stack when it comes to includes, which
+ means that a context can only contain strictly fewer than
+ AST_PBX_MAX_STACK includes. If this is exceeded, then warnings
+ will be emitted for each number of includes beyond this if
+ searching for an extension in the including context, and if
+ the extension's inclusion is beyond the stack size, it will
+ simply not be found.
+
+ To address this, we now check if there are too many includes
+ in a context when the dialplan is reloaded so that if there
+ is an issue, the user is aware of at "compile time" as opposed
+ to "run time" only. Secondly, more details are printed out
+ when this message is encountered so it's clear what has happened.
+
+ ASTERISK-26719
+
+
+#### Makefile: Disable XML doc validation
+ Author: George Joseph
+ Date: 2022-03-25
+
+ make_xml_documentation was being called with the --validate
+ flag set when it shouldn't have been. This was causing
+ build failures if neither xmllint nor xmlstarlet were installed.
+ The correct behavior is to simply print a message that either
+ one of those tools should be installed for validation and
+ continue with the build.
+
+ ASTERISK-29988
+
+
+#### make_xml_documentation: Remove usage of get_sourceable_makeopts
+ Author: George Joseph
+ Date: 2022-03-25
+
+ get_sourceable_makeopts wasn't handling variables with embedded
+ double quotes in them very well. One example was the DOWNLOAD
+ variable when curl was being used instead of wget. Rather than
+ trying to fix get_sourceable_makeopts, it's just been removed.
+
+ ASTERISK-29986
+ Reported by: Stefan Ruijsenaars
+
+
+#### chan_iax2: Fix spacing in netstats command
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ The iax2 show netstats command previously didn't contain
+ enough spacing in the header to properly align the table
+ header with the table body. This caused column headers
+ to not align with the values on longer channel names.
+
+ Some spacing is added to account for the longest channel
+ names that display (before truncation occurs) so that
+ columns are always properly aligned.
+
+ ASTERISK-29895 #close
+ patches:
+ 61205_misaligned2.patch submitted by Birger Harzenetter (license 5870)
+
+
+#### openssl: Supress deprecation warnings from OpenSSL 3.0
+ Author: Sean Bright
+ Date: 2022-03-25
+
+ There is work going on to update our OpenSSL usage to avoid the
+ deprecated functions but in the meantime make it possible to compile
+ in devmode.
+
+
+#### documentation: Add information on running install_prereq script in readme
+ Author: Marcel Wagner
+ Date: 2022-03-23
+
+ Adding information in the readme about running the install_preqreq script to install components that the ./configure script might indicate as missing.
+
+ ASTERISK-29976 #close
+
+
+#### chan_iax2: Fix perceived showing host address.
+ Author: Naveen Albert
+ Date: 2022-03-13
+
+ ASTERISK_22025 introduced a regression that shows
+ the host IP and port as the perceived IP and port
+ again, as opposed to showing the actual perceived
+ address. This fixes this by showing the correct
+ information.
+
+ ASTERISK-29048 #close
+
+
+#### res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity
+ Author: Boris P. Korzun
+ Date: 2022-02-22
+
+ Change RTP timer behavior for detecting RTP only after two-way
+ SDP channel establishment. Ignore detecting after receiving 183
+ with SDP or while direct media is used.
+ Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
+ and rtpholdtimeout options in chan_sip.
+
+ ASTERISK-26689 #close
+ ASTERISK-29929 #close
+
+
+#### configure.ac: Use pkg-config to detect libxml2
+ Author: Hugh McMaster
+ Date: 2022-03-16
+
+ Use pkg-config to detect libxml2, falling back to xml2-config if the
+ former is not available.
+
+ This patch ensures Asterisk continues to build on systems without
+ xml2-config installed.
+
+ The patch also updates the associated 'configure' files.
+
+ ASTERISK-29970 #close
+
+
+#### time: add support for time64 libcs
+ Author: Philip Prindeville
+ Date: 2022-02-13
+
+ Treat time_t's as entirely unique and use the POSIX API's for
+ converting to/from strings.
+
+ Lastly, a 64-bit integer formats as 20 digits at most in base10.
+ Don't need to have any 100 byte buffers to hold that.
+
+ ASTERISK-29674 #close
+
+ Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
+
+#### res_pjsip_pubsub: RLS 'uri' list attribute mismatch with SUBSCRIBE request
+ Author: Alexei Gradinari
+ Date: 2022-03-15
+
+ When asterisk generates the RLMI part of NOTIFY request,
+ the asterisk uses the local contact uri instead of the URI to which
+ the SUBSCRIBE request is sent.
+ Because of this mismatch some IP phones (for example Cisco 5XX) ignore
+ this list.
+
+ According
+ https://datatracker.ietf.org/doc/html/rfc4662#section-5.2
+ The first mandatory <list> attribute is "uri", which contains the uri
+ that corresponds to the list. Typically, this is the URI to which
+ the SUBSCRIBE request was sent.
+ https://datatracker.ietf.org/doc/html/rfc4662#section-5.3
+ The "uri" attribute identifies the resource to which the <resource>
+ element corresponds. Typically, this will be a SIP URI that, if
+ subscribed to, would return the state of the resource.
+
+ This patch makes asterisk to generate URI using SUBSCRIBE request URI.
+
+ ASTERISK-29961 #close
+
+
+#### app_dial: Document DIALSTATUS return values.
+ Author: Naveen Albert
+ Date: 2022-03-05
+
+ Adds documentation for all of the possible return values
+ for the DIALSTATUS variable in the Dial application.
+
+ ASTERISK-25716
+
+
+#### stasis_recording: Perform a complete match on requested filename.
+ Author: Sean Bright
+ Date: 2022-03-10
+
+ Using the length of a file found on the filesystem rather than the
+ file being requested could result in filenames whose names are
+ substrings of another to be erroneously matched.
+
+ We now ensure a complete comparison before returning a positive
+ result.
+
+ ASTERISK-29960 #close
+
+
+#### download_externals: Use HTTPS for downloads
+ Author: Sean Bright
+ Date: 2022-03-22
+
+ ASTERISK-29980 #close
+
+
+#### conversions.c: Specify that we only want to parse decimal numbers.
+ Author: Sean Bright
+ Date: 2022-03-04
+
+ Passing 0 as the last argument to strtoimax() or strtoumax() causes
+ octal and hexadecimal to be accepted which was not originally
+ intended. So we now force to only accept decimal.
+
+ ASTERISK-29950 #close
+
+
+#### logger: workaround woefully small BUFSIZ in MUSL
+ Author: Philip Prindeville
+ Date: 2022-02-21
+
+ MUSL defines BUFSIZ as 1024 which is not reasonable for log messages.
+
+ More broadly, BUFSIZ is the amount of buffering stdio.h does, which
+ is arbitrary and largely orthogonal to what logging should accept
+ as the maximum message size.
+
+ ASTERISK-29928
+
+ Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
+
+#### pbx_builtins: Add missing options documentation
+ Author: Naveen Albert
+ Date: 2022-03-14
+
+ BackGround and WaitExten both accept options that are not
+ currently documented. This adds documentation for these
+ options to the xml documentation for each application.
+
+ ASTERISK-29967 #close
+
+
+#### res_pjsip_pubsub: update RLS to reflect the changes to the lists
+ Author: Alexei Gradinari
+ Date: 2022-02-08
+
+ This patch makes the Resource List Subscriptions (RLS) dynamic.
+ The asterisk updates the current subscriptions to reflect the changes
+ to the list on the subscriptions refresh. If list items are added,
+ removed, updated or do not exist anymore, the asterisk regenerates
+ the resource list.
+
+ ASTERISK-29906 #close
+
+
+#### res_agi: Fix xmldocs bug with set music.
+ Author: Naveen Albert
+ Date: 2022-02-25
+
+ The XML documentation for the SET MUSIC AGI
+ command is invalid, as the parameter does not
+ have a name and the on/off enum options for
+ the on/off argument are listed separately, which
+ is incorrect. The cumulative effect of these currently
+ is that the Asterisk Wiki documentation for SET MUSIC
+ is broken and external documentation generators crash
+ on SET MUSIC due to the malformed documentation.
+
+ These issues are corrected so that the documentation
+ can be successfully parsed as with other similar AGI
+ commands.
+
+ ASTERISK-29939 #close
+ ASTERISK-28891 #close
+
+
+#### res_config_pgsql: Add text-type column check in require_pgsql()
+ Author: Boris P. Korzun
+ Date: 2022-02-18
+
+ Omit "unsupported column type 'text'" warning in logs while
+ using text-type column in the PgSQL backend.
+
+ ASTERISK-29924 #close
+
+
+#### app_queue: Add QueueWithdrawCaller AMI action
+ Author: Kfir Itzhak
+ Date: 2022-02-09
+
+ This adds a new AMI action called QueueWithdrawCaller.
+ This AMI action makes it possible to withdraw a caller from a queue,
+ in a safe and a generic manner.
+ This can be useful for retrieving a specific call and
+ dispatching it to a specific extension.
+ It works by signaling the caller to exit the queue application
+ whenever it can. Therefore, it is not guaranteed
+ that the call will leave the queue.
+
+ ASTERISK-29909 #close
+
+
+#### ami: Improve substring parsing for disabled events.
+ Author: Naveen Albert
+ Date: 2022-02-24
+
+ ASTERISK_29853 added the ability to selectively disable
+ AMI events on a global basis, but the logic for this uses
+ strstr which means that events with names which are the prefix
+ of another event, if disabled, could disable those events as
+ well.
+
+ Instead, we account for this possibility to prevent this
+ undesired behavior from occuring.
+
+ ASTERISK_29853
+
+
+#### xml.c, config,c: Add stylesheets and variable list string parsing
+ Author: George Joseph
+ Date: 2022-03-02
+
+ Added functions to open, close, and apply XML Stylesheets
+ to XML documents. Although the presence of libxslt was already
+ being checked by configure, it was only happening if xmldoc was
+ enabled. Now it's checked regardless.
+
+ Added ability to parse a string consisting of comma separated
+ name/value pairs into an ast_variable list. The reverse of
+ ast_variable_list_join().
+
+
+#### xmldoc: Fix issue with xmlstarlet validation
+ Author: George Joseph
+ Date: 2022-03-01
+
+ Added the missing xml-stylesheet and Xinclude namespace
+ declarations in pjsip_config.xml and pjsip_manager.xml.
+
+ Updated make_xml_documentation to show detailed errors when
+ xmlstarlet is the validator. It's now run once with the '-q'
+ option to suppress harmless/expected messages and if it actually
+ fails, it's run again without '-q' but with '-e' to show
+ the actual errors.
+
+
+#### core: Config and XML tweaks needed for geolocation
+ Author: George Joseph
+ Date: 2022-02-20
+
+ Added:
+
+ Replace a variable in a list:
+ int ast_variable_list_replace_variable(struct ast_variable **head,
+ struct ast_variable *old, struct ast_variable *new);
+ Added test as well.
+
+ Create a "name=value" string from a variable list:
+ 'name1="val1",name2="val2"', etc.
+ struct ast_str *ast_variable_list_join(
+ const struct ast_variable *head, const char *item_separator,
+ const char *name_value_separator, const char *quote_char,
+ struct ast_str **str);
+ Added test as well.
+
+ Allow the name of an XML element to be changed.
+ void ast_xml_set_name(struct ast_xml_node *node, const char *name);
+
+
+#### Makefile: Allow XML documentation to exist outside source files
+ Author: George Joseph
+ Date: 2022-02-14
+
+ Moved the xmldoc build logic from the top-level Makefile into
+ its own script "make_xml_documentation" in the build_tools
+ directory.
+
+ Created a new utility script "get_sourceable_makeopts", also in
+ the build_tools directory, that dumps the top-level "makeopts"
+ file in a format that can be "sourced" from shell sscripts.
+ This allows scripts to easily get the values of common make
+ build variables such as the location of the GREP, SED, AWK, etc.
+ utilities as well as the AST* and library *_LIB and *_INCLUDE
+ variables.
+
+ Besides moving logic out of the Makefile, some optimizations
+ were done like removing "third-party" from the list of
+ subdirectories to be searched for documentation and changing some
+ assignments from "=" to ":=" so they're only evaluated once.
+ The speed increase is noticeable.
+
+ The makeopts.in file was updated to include the paths to
+ REALPATH and DIRNAME. The ./conifgure script was setting them
+ but makeopts.in wasn't including them.
+
+ So...
+
+ With this change, you can now place documentation in any"c"
+ source file AND you can now place it in a separate XML file
+ altogether. The following are examples of valid locations:
+
+ res/res_pjsip.c
+ Using the existing /*** DOCUMENTATION ***/ fragment.
+
+ res/res_pjsip/pjsip_configuration.c
+ Using the existing /*** DOCUMENTATION ***/ fragment.
+
+ res/res_pjsip/pjsip_doc.xml
+ A fully-formed XML file. The "configInfo", "manager",
+ "managerEvent", etc. elements that would be in the "c"
+ file DOCUMENTATION fragment should be wrapped in proper
+ XML. Example for "somemodule.xml":
+
+ <?xml version="1.0" encoding="UTF-8"?>
+ <!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+ <docs>
+ <configInfo>
+ ...
+ </configInfo>
+ </docs>
+
+ It's the "appdocsxml.dtd" that tells make_xml_documentation
+ that this is a documentation XML file and not some other XML file.
+ It also allows many XML-capable editors to do formatting and
+ validation.
+
+ Other than the ".xml" suffix, the name of the file is not
+ significant.
+
+ As a start... This change also moves the documentation that was
+ in res_pjsip.c to 2 new XML files in res/res_pjsip:
+ pjsip_config.xml and pjsip_manager.xml. This cut the number of
+ lines in res_pjsip.c in half. :)
+
+
+#### build: Refactor the earlier "basebranch" commit
+ Author: George Joseph
+ Date: 2022-02-17
+
+ Recap from earlier commit: If you have a development branch for a
+ major project that will receive gerrit reviews it'll probably be
+ named something like "development/16/newproject" or a work branch
+ based on that "development" branch. That will necessitate
+ setting "defaultbranch=development/16/newproject" in .gitreview.
+ The make_version script uses that variable to construct the
+ asterisk version however, which results in versions
+ like "GIT-development/16/newproject-ee582a8c7b" which is probably
+ not what you want. It also constructs the URLs for downloading
+ external modules with that version, which will fail.
+
+ Fast-forward:
+
+ The earlier attempt at adding a "basebranch" variable to
+ .gitreview didn't work out too well in practice because changes
+ were made to .gitreview, which is a checked-in file. So, if
+ you wanted to rebase your work branch on the base branch, rebase
+ would attempt to overwrite your .gitreview with the one from
+ the base branch and complain about a conflict.
+
+ This is a slighltly different approach that adds three methods to
+ determine the mainline branch:
+
+ 1. --- MAINLINE_BRANCH from the environment
+
+ If MAINLINE_BRANCH is already set in the environment, that will
+ be used. This is primarily for the Jenkins jobs.
+
+ 2. --- .develvars
+
+ Instead of storing the basebranch in .gitreview, it can now be
+ stored in a non-checked-in ".develvars" file and keyed by the
+ current branch. So, if you were working on a branch named
+ "new-feature-work" based on "development/16/new-feature" and wanted
+ to push to that branch in Gerrit but wanted to pull the external
+ modules for 16, you'd create the following .develvars file:
+
+ [branch "new-feature-work"]
+ mainline-branch = 16
+
+ The .gitreview file would still look like:
+
+ [gerrit]
+ defaultbranch=development/16/new-feature
+
+ ...which would cause any reviews pushed from "new-feature-work" to
+ go to the "development/16/new-feature" branch in Gerrit.
+
+ The key is that the .develvars file is NEVER checked in (it's been
+ added to .gitignore).
+
+ 3. --- Well Known Development Branch
+
+ If you're actually working in a branch named like
+ "development/<mainline_branch>/some-feature", the mainline branch
+ will be parsed from it.
+
+ 4. --- .gitreview
+
+ If none of the earlier conditions exist, the .gitreview
+ "defaultbranch" variable will be used just as before.
+
+
+#### jansson: Update bundled to 2.14 version.
+ Author: Joshua C. Colp
+ Date: 2022-02-23
+
+ ASTERISK-29353
+
+
+#### func_channel: Add lastcontext and lastexten.
+ Author: Naveen Albert
+ Date: 2022-01-06
+
+ Adds the lastcontext and lastexten channel fields to allow users
+ to access previous dialplan execution locations.
+
+ ASTERISK-29840 #close
+
+
+#### channel.c: Clean up debug level 1.
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ Although there are 10 debugs levels, over time,
+ many current debug calls have come to use
+ inappropriately low debug levels. In particular,
+ a select few debug calls (currently all debug 1)
+ can result in thousands of debug messages per minute
+ for a single call.
+
+ This can adds a lot of noise to core debug
+ which dilutes the value in having different
+ debug levels in the first place, as these
+ log messages are from the core internals are
+ are better suited for higher debug levels.
+
+ Some debugs levels are thus adjusted so that
+ debug level 1 is not inappropriately overloaded
+ with these extremely high-volume and general
+ debug messages.
+
+ ASTERISK-29897 #close
+
+
+#### configs, LICENSE: remove pbx.digium.com.
+ Author: Naveen Albert
+ Date: 2022-02-17
+
+ pbx.digium.com no longer accepts IAX2 calls and
+ there are no plans for it to come back.
+
+ Accordingly, nonworking IAX2 URIs are removed from
+ both the LICENSE file and the sample config.
+
+ ASTERISK-29923 #close
+
+
+#### documentation: Add since tag to xmldocs DTD
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ Adds the since tag to the documentation DTD so
+ that individual applications, functions, etc.
+ can now specify when they were added to Asterisk.
+
+ This tag is added at the individual application,
+ function, etc. level as opposed to at the module
+ level because modules can expand over time as new
+ functionality is added, and granularity only
+ to the module level would generally not be useful.
+
+ This enables the ability to more easily determine
+ when new functionality was added to Asterisk, down
+ to minor version as opposed to just by major version.
+ This makes it easier for users to write more portable
+ dialplan if desired to not use functionality that may
+ not be widely available yet.
+
+ ASTERISK-29896 #close
+
+
+#### asterisk: Add macro for curl user agent.
+ Author: Naveen Albert
+ Date: 2022-01-13
+
+ Currently, each module that uses libcurl duplicates the standard
+ Asterisk curl user agent.
+
+ This adds a global macro for the Asterisk user agent used for
+ curl requests to eliminate this duplication.
+
+ ASTERISK-29861 #close
+
+
+#### res_stir_shaken: refactor utility function
+ Author: Naveen Albert
+ Date: 2021-12-16
+
+ Refactors temp file utility function into file.c.
+
+ ASTERISK-29809 #close
+
+
+#### app_voicemail: Emit warning if asking for nonexistent mailbox.
+ Author: Naveen Albert
+ Date: 2022-02-16
+
+ Currently, if VoiceMailMain is called with a mailbox, if that
+ mailbox doesn't exist, then the application silently falls back
+ to prompting the user for the mailbox, as if no arguments were
+ provided.
+
+ However, if a specific mailbox is requested and it doesn't exist,
+ then no warning at all is emitted.
+
+ This fixes this behavior to now warn if a specifically
+ requested mailbox could not be accessed, before falling back to
+ prompting the user for the correct mailbox.
+
+ ASTERISK-29920 #close
+
+
+#### res_pjsip_pubsub: fix Batched Notifications stop working
+ Author: Alexei Gradinari
+ Date: 2022-02-07
+
+ If Subscription refresh occurred between when the batched notification
+ was scheduled and the serialized notification was to be sent,
+ then new schedule notification task would never be added.
+
+ There are 2 threads:
+
+ thread #1. ast_sip_subscription_notify is called,
+ if notification_batch_interval then call schedule_notification.
+ 1.1. The schedule_notification checks notify_sched_id > -1
+ not true, then
+ send_scheduled_notify = 1
+ notify_sched_id =
+ ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
+ 1.2. The sched_cb pushes task serialized_send_notify to serializer
+ and returns 0 which means no reschedule.
+ 1.3. The serialized_send_notify checks send_scheduled_notify if it's false
+ the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.
+
+ thread #2. pubsub_on_rx_refresh is called
+ 2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
+ 2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
+ which calls send_notify
+ 2.3. The send_notify set send_scheduled_notify = 0;
+
+ The serialized_send_notify should always unset notify_sched_id.
+
+ ASTERISK-29904 #close
+
+
+#### res_pjsip_pubsub: provide a display name for RLS subscriptions
+ Author: Alexei Gradinari
+ Date: 2022-02-01
+
+ Whereas BLFs allow to show a display name for each RLS entry,
+ the asterisk provides only the extension now.
+ This is not end user friendly.
+
+ This commit adds a new resource_list option, resource_display_name,
+ to indicate whether display name of resource or the resource name being
+ provided for RLS entries.
+ If this option is enabled, the Display Name will be provided.
+ This option is disabled by default to remain the previous behavior.
+ If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
+ will be set as the Display Name.
+ The 'message-summary' is not supported yet.
+
+ ASTERISK-29891 #close
+
+
+#### func_db: Add validity check for key names when writing.
+ Author: Naveen Albert
+ Date: 2022-02-18
+
+ Adds a simple sanity check for key names when users are
+ writing data to AstDB. This captures four cases indicating
+ malformed keynames that generally result in bad data going
+ into the DB that the user didn't intend: an empty key name,
+ a key name beginning or ending with a slash, and a key name
+ containing two slashes in a row. Generally, this is the
+ result of a variable being used in the key name being empty.
+
+ If a malformed key name is detected, a warning is emitted
+ to indicate the bug in the dialplan.
+
+ ASTERISK-29925 #close
+
+
+#### cli: Add core dump info to core show settings.
+ Author: Naveen Albert
+ Date: 2022-01-14
+
+ Adds two pieces of information to the core show settings command
+ which are useful in the context of getting backtraces.
+
+ The first is to display whether or not Asterisk would generate
+ a core dump if it were to crash.
+
+ The second is to show the current running directory of Asterisk.
+
+ ASTERISK-29866 #close
+
+
+#### documentation: Adds missing default attributes.
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ The configObject tag contains a default attribute which
+ allows the default value to be specified, if applicable.
+ This allows for the default value to show up specially on
+ the wiki in a way that is clear to users.
+
+ There are a couple places in the tree where default values
+ are included in the description as opposed to as attributes,
+ which means these can't be parsed specially for the wiki.
+ These are changed to use the attribute instead of being
+ included in the text description.
+
+ ASTERISK-29898 #close
+
+
+#### app_mp3: Document and warn about HTTPS incompatibility.
+ Author: Naveen Albert
+ Date: 2022-02-05
+
+ mpg123 doesn't support HTTPS, but the MP3Player application
+ doesn't document this or warn the user about this. HTTPS
+ streams have become more common nowadays and users could
+ reasonably try to play them without being aware they should
+ use the HTTP stream instead.
+
+ This adds documentation to note this limitation. It also
+ throws a warning if users try to use the HTTPS stream to
+ tell them to use the HTTP stream instead.
+
+ ASTERISK-29900 #close
+
+
+#### app_mf: Add max digits option to ReceiveMF.
+ Author: Naveen Albert
+ Date: 2022-01-22
+
+ Adds an option to the ReceiveMF application to allow specifying a
+ maximum number of digits.
+
+ Originally, this capability was not added to ReceiveMF as it was
+ with ReceiveSF because typically a ST digit is used to denote that
+ sending of digits is complete. However, there are certain signaling
+ protocols which simply transmit a digit (such as Expanded In-Band
+ Signaling) and for these, it's necessary to be able to read a
+ certain number of digits, as opposed to until receiving a ST digit.
+
+ This capability is added as an option, as opposed to as a parameter,
+ to remain compatible with existing usage (and not shift the
+ parameters).
+
+ ASTERISK-29877 #close
+
+
+#### ami: Allow events to be globally disabled.
+ Author: Naveen Albert
+ Date: 2022-01-09
+
+ The disabledevents setting has been added to the general section
+ in manager.conf, which allows users to specify events that
+ should be globally disabled and not sent to any AMI listeners.
+
+ This allows for processing of these AMI events to end sooner and,
+ for frequent AMI events such as Newexten which users may not have
+ any need for, allows them to not be processed. Additionally, it also
+ cleans up core debug as previously when debug was 3 or higher,
+ the debug was constantly spammed by "Analyzing AMI event" messages
+ along with a complete dump of the event contents (often for Newexten).
+
+ ASTERISK-29853 #close
+
+
+#### taskprocessor.c: Prevent crash on graceful shutdown
+ Author: Mike Bradeen
+ Date: 2022-02-02
+
+ When tps_shutdown is called as part of the cleanup process there is a
+ chance that one of the taskprocessors that references the
+ tps_singletons object is still running. The change is to allow for
+ tps_shutdown to check tps_singleton's container count and give the
+ running taskprocessors a chance to finish. If after
+ AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still
+ container references we shutdown anyway as this is most likely a bug
+ due to a taskprocessor not being unreferenced.
+
+ ASTERISK-29365
+
+
+#### app_queue: load queues and members from Realtime when needed
+ Author: Alexei Gradinari
+ Date: 2022-01-21
+
+ There are a lot of Queue AMI actions and Queue applications
+ which do not load queue and queue members from Realtime.
+
+ AMI actions
+ QueuePause - if queue not in memory - response "Interface not found".
+ QueueStatus/QueueSummary - if queue not in memory - empty response.
+
+ Applications:
+ PauseQueueMember - if queue not in memory
+ Attempt to pause interface %s, not found
+ UnpauseQueueMember - if queue not in memory
+ Attempt to unpause interface xxxxx, not found
+
+ This patch adds a new function load_realtime_queues
+ which loads queue and queue members for desired queue
+ or all queues and all members if param 'queuename' is NULL or empty.
+ Calls the function load_realtime_queues when needed.
+
+ Also this patch fixes leak of ast_config in function set_member_value.
+
+ Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
+ already paused/unpaused member.
+ The function ast_update_realtime returns 0 when no record modified.
+ So 0 is not an error to warn about.
+
+ ASTERISK-29873 #close
+ ASTERISK-18416 #close
+ ASTERISK-27597 #close
+
+
+#### manager.c: Simplify AMI ModuleCheck handling
+ Author: Sean Bright
+ Date: 2022-02-07
+
+ This code was needlessly complex and would fail to properly delimit
+ the response message if LOW_MEMORY was defined.
+
+
+#### res_prometheus.c: missing module dependency
+ Author: Mark Petersen
+ Date: 2022-01-21
+
+ added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO
+ which fixes issue with module crashes on load "FRACK!, Failed assertion"
+
+ ASTERISK-29871
+
+
+#### res_pjsip.c: Correct minor typos in 'realm' documentation.
+ Author: Sean Bright
+ Date: 2022-02-03
+
+
+#### manager.c: Generate valid XML if attribute names have leading digits.
+ Author: Sean Bright
+ Date: 2022-01-31
+
+ The XML Manager Event Interface (amxml) now generates attribute names
+ that are compliant with the XML 1.1 specification. Previously, an
+ attribute name that started with a digit would be rendered as-is, even
+ though attribute names must not begin with a digit. We now prefix
+ attribute names that start with a digit with an underscore ('_') to
+ prevent XML validation failures.
+
+ This is not backwards compatible but my assumption is that compliant
+ XML parsers would already have been complaining about this.
+
+ ASTERISK-29886 #close
+
+
+#### build_tools/make_version: Fix bashism in comparison.
+ Author: Sean Bright
+ Date: 2022-02-01
+
+ In POSIX sh (which we indicate in the shebang), there is no ==
+ operator.
+
+
+#### bundled_pjproject: Add additional multipart search utils
+ Author: George Joseph
+ Date: 2022-01-21
+
+ Added the following APIs:
+ pjsip_multipart_find_part_by_header()
+ pjsip_multipart_find_part_by_header_str()
+ pjsip_multipart_find_part_by_cid_str()
+ pjsip_multipart_find_part_by_cid_uri()
+
+
+#### chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN
+ Author: Mark Petersen
+ Date: 2022-01-07
+
+ resolve issue with pickup on device that uses "183" and not "180"
+
+ ASTERISK-29832
+
+
+#### build: Add "basebranch" to .gitreview
+ Author: George Joseph
+ Date: 2022-01-26
+
+ If you have a development branch for a major project that
+ will receive gerrit reviews it'll probably be named something
+ like "development/16/newproject". That will necessitate setting
+ "defaultbranch=development/16/newproject" in .gitreview. The
+ make_version script uses that variable to construct the asterisk
+ version however, which results in versions like
+ "GIT-development/16/newproject-ee582a8c7b" which is probably not
+ what you want. Worse, since the download_externals script uses
+ make_version to construct the URL to download the binary codecs
+ or DPMA. Since it's expecting a simple numeric version, the
+ downloads will fail.
+
+ To get this to work, a new variable "basebranch" has been added
+ to .gitreview and make_version has been updated to use that instead
+ of defaultversion:
+
+ .gitreview:
+ defaultbranch=development/16/myproject
+ basebranch=16
+
+ Now git-review will send the reviews to the proper branch
+ (development/16/myproject) but the version will still be
+ constructed using the simple branch number (16).
+
+ If "basebranch" is missing from .gitreview, make_version will
+ fall back to using "defaultbranch".
+
+
+#### res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup
+ Author: George Joseph
+ Date: 2022-01-31
+
+ In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess
+ that hasn't been initialized, it'll assert and abort. If
+ digest_create_request_with_auth() fails to find the proper
+ auth object however, it jumps to its cleanup which does exactly
+ that. So now we no longer attempt to call pjsip_auth_clt_deinit()
+ if we never actually initialized it.
+
+ ASTERISK-29888
+
+
+#### cdr: allow disabling CDR by default on new channels
+ Author: Naveen Albert
+ Date: 2021-12-15
+
+ Adds a new option, defaultenabled, to the CDR core to
+ control whether or not CDR is enabled on a newly created
+ channel. This allows CDR to be disabled by default on
+ new channels and require the user to explicitly enable
+ CDR if desired. Existing behavior remains unchanged.
+
+ ASTERISK-29808 #close
+
+
+#### res_tonedetect: Fixes some logic issues and typos
+ Author: Naveen Albert
+ Date: 2022-01-11
+
+ Fixes some minor logic issues with the module:
+
+ Previously, the OPT_END_FILTER flag was getting
+ tested before options were parsed, so it could
+ never evaluate to true (wrong ordering).
+
+ Additionally, the initially parsed timeout (float)
+ needs to be compared with 0, not the result int
+ which is set afterwards (wrong variable).
+
+ ASTERISK-29857 #close
+
+
+#### func_frame_drop: Fix typo referencing wrong buffer
+ Author: Naveen Albert
+ Date: 2022-01-11
+
+ In order to get around the issue of certain frames
+ having names that could overlap, func_frame_drop
+ surrounds names with commas for the purposes of
+ comparison.
+
+ The buffer is allocated and printed to properly,
+ but the original buffer is used for comparison.
+ In most cases, this wouldn't have had any effect,
+ but that was not the intention behind the buffer.
+ This updates the code to reference the modified
+ buffer instead.
+
+ ASTERISK-29854 #close
+
+
+#### res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf
+ Author: Torrey Searle
+ Date: 2022-01-20
+
+ When generating dtmfs, asterisk can incorrectly think packet loss
+ occured during the dtmf generation, resulting in a jump in sequence
+ numbers when forwarding voice frames resumes. This patch forces
+ asterisk to re-learn the expected sequence number after each DTMF
+ to avoid this
+
+ ASTERISK-29869 #close
+
+
+#### res_http_websocket: Add a client connection timeout
+ Author: Kevin Harwell
+ Date: 2022-01-13
+
+ Previously there was no way to specify a connection timeout when
+ attempting to connect a websocket client to a server. This patch
+ makes it possible to now do such.
+
+
+#### build: Rebuild configure and autoconfig.h.in
+ Author: Sean Bright
+ Date: 2022-01-21
+
+ autoconfigh.h.in was missed in the original review for this
+ issue. Additionally it looks like I have newer pkg-config autoconf
+ macros on my development machine.
+
+ ASTERISK-29817
+
+
+#### sched: fix and test a double deref on delete of an executing call back
+ Author: Mike Bradeen
+ Date: 2021-12-08
+
+ sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an
+ executing call-back. This is done by adding a new variable 'rescheduled'
+ to the struct sched which is set in ast_sched_runq and checked in
+ ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for
+ now deprecated ast_sched_del which returns a new possible value -2
+ if called on an executing call-back with rescheduled set. ast_sched_del
+ is modified to call ast_sched_del_nonrunning to maintain existing code.
+ AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it
+ will not throw a warning or invoke refcall.
+ test_sched: Add a new unit test sched_test_freebird that will check the
+ reference count in the resolved scenario.
+
+ ASTERISK-29698
+
+
+#### app_queue.c: Queue don't play "thank-you" when here is no hold time announceme..
+ Author: Mark Petersen
+ Date: 2022-01-04
+
+ if holdtime is (0 min, 0 sec) there is no hold time announcements
+ we should then also not playing queue-thankyou
+
+ ASTERISK-29831
+
+
+#### res_pjsip_sdp_rtp.c: Support keepalive for video streams.
+ Author: Luke Escude
+ Date: 2022-01-19
+
+ ASTERISK-28890 #close
+
+
+#### build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD
+ Author: Michał Górny
+ Date: 2021-11-11
+
+ Fix the sed(1) invocation used to process git-svn-id not to use "\s"
+ that is a GNU-ism and is not supported by NetBSD sed. As a result,
+ this call did not work properly and make_version did output the full
+ git-svn-id line rather than the revision.
+
+ ASTERISK-29852
+
+
+#### main/utils: Implement ast_get_tid() for NetBSD
+ Author: Michał Górny
+ Date: 2021-11-11
+
+ Implement the ast_get_tid() function for NetBSD system. NetBSD supports
+ getting the TID via _lwp_self().
+
+ ASTERISK-29850
+
+
+#### main: Enable rdtsc support on NetBSD
+ Author: Michał Górny
+ Date: 2021-11-11
+
+ Enable the Linux rdtsc implementation on NetBSD as well. The assembly
+ works correctly there.
+
+ ASTERISK-29851
+
+
+#### BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD
+ Author: Michał Górny
+ Date: 2021-11-11
+
+ Fix the configure script not to detect the presence of gethostbyname_r()
+ on NetBSD incorrectly. NetBSD includes it as an internal libc symbol
+ that is not exposed in system headers and that is incompatible with
+ other implementations. In order to avoid misdetecting it, perform
+ the symbol check only if the declaration is found in the public header
+ first.
+
+ ASTERISK-29817
+
+
+#### include: Remove unimplemented HMAC declarations
+ Author: Michał Górny
+ Date: 2021-11-11
+
+ Remove the HMAC declarations from the includes. They are
+ not implemented nor used anywhere, and their presence breaks the build
+ on NetBSD that delivers an incompatible hmac() function in <stdlib.h>.
+
+ ASTERISK-29818
+
+
+#### frame.h: Fix spelling typo
+ Author: Naveen Albert
+ Date: 2022-01-11
+
+ Fixes CNG description from "noice" to "noise".
+
+ ASTERISK-29855 #close
+
+
+#### res_rtp_asterisk: Fix typo in flag test/set
+ Author: Naveen Albert
+ Date: 2022-01-11
+
+ The code currently checks to see if an RFC3389
+ warning flag is set, except if it is, it merely
+ sets the flag again, the logic of which doesn't
+ make any sense.
+
+ This adjusts the if comparison to check if the
+ flag has NOT been set, and if so, emit a notice
+ log event and set the flag so that future frames
+ do not cause an event to be logged.
+
+ ASTERISK-29856 #close
+
+
+#### bundled_pjproject: Fix srtp detection
+ Author: George Joseph
+ Date: 2022-01-18
+
+ Reverted recent change that set '--with-external-srtp' instead
+ of '--without-external-srtp'. Since Asterisk handles all SRTP,
+ we don't need it enabled in pjproject at all.
+
+ ASTERISK-29867
+
+
+#### res_pjsip: Make message_filter and session multipart aware
+ Author: George Joseph
+ Date: 2022-01-10
+
+ Neither pjsip_message_filter's filter_on_tx_message() nor
+ res_pjsip_session's session_outgoing_nat_hook() were multipart
+ aware and just assumed that an SDP would be the only thing in
+ a message body. Both were changed to use the new
+ pjsip_get_sdp_info() function which searches for an sdp in
+ both single- and multi- part message bodies.
+
+ ASTERISK-29813
+
+
+#### build: Fix issues building pjproject
+ Author: George Joseph
+ Date: 2022-01-12
+
+ The change to allow easier hacking on bundled pjproject created
+ a few issues:
+
+ * The new Makefile was trying to run the bundled make even if
+ PJPROJECT_BUNDLED=no. third-party/Makefile now checks for
+ PJPROJECT_BUNDLED and JANSSON_BUNDLED and skips them if they
+ are "no".
+
+ * When building with bundled, config_site.h was being copied
+ only if a full make or a "make main" was done. A "make res"
+ would fail all the pjsip modules because they couldn't find
+ config_site.h. The Makefile now copies config_site.h and
+ asterisk_malloc_debug.h into the pjproject source tree
+ when it's "configure" is performed. This is how it used
+ to be before the big change.
+
+ ASTERISK-29858
+
+
+#### res_pjsip: Add utils for checking media types
+ Author: George Joseph
+ Date: 2022-01-06
+
+ Added two new functions to assist checking media types...
+
+ * ast_sip_are_media_types_equal compares two pjsip_media_types.
+ * ast_sip_is_media_type_in tests if one media type is in a list
+ of others.
+
+ Added static definitions for commonly used media types to
+ res_pjsip.h.
+
+ Changed several modules to use the new functions and static
+ definitions.
+
+ ASTERISK_29813
+ (not ready to close)
+
+
+#### bundled_pjproject: Create generic pjsip_hdr_find functions
+ Author: George Joseph
+ Date: 2022-01-12
+
+ pjsip_msg_find_hdr(), pjsip_msg_find_hdr_by_name(), and
+ pjsip_msg_find_hdr_by_names() require a pjsip_msg to be passed in
+ so if you need to search a header list that's not in a pjsip_msg,
+ you have to do it yourself. This commit adds generic versions of
+ those 3 functions that take in the actual header list head instead
+ of a pjsip_msg so if you need to search a list of headers in
+ something like a pjsip_multipart_part, you can do so easily.
+
+
+#### say.c: Prevent erroneous failures with 'say' family of functions.
+ Author: Sean Bright
+ Date: 2022-01-12
+
+ A regression was introduced in ASTERISK~29531 that caused 'say'
+ functions to fail with file lists that would previously have
+ succeeded. This caused affected channels to hang up where previously
+ they would have continued.
+
+ We now explicitly check for the empty string to restore the previous
+ behavior.
+
+ ASTERISK-29859 #close
+
+
+#### documentation: Document built-in system and channel vars
+ Author: Naveen Albert
+ Date: 2022-01-08
+
+ Documentation for built-in special system and channel
+ vars is currently outdated, and updating is a manual
+ process since there is no XML documentation for these
+ anywhere.
+
+ This adds documentation for system vars to func_env
+ and for channel vars to func_channel so that they
+ appear along with the corresponding fields that would
+ be accessed using a function.
+
+ ASTERISK-29848 #close
+
+
+#### pbx_variables: add missing ASTSBINDIR variable
+ Author: Naveen Albert
+ Date: 2022-01-08
+
+ Every config variable in the directories
+ section of asterisk.conf currently has a
+ counterpart built-in variable containing
+ the value of the config option, except
+ for the last one, astsbindir, which should
+ have an ASTSBINDIR variable.
+
+ However, the actual corresponding ASTSBINDIR
+ variable is missing in pbx_variables.c.
+
+ This adds the missing variable so that all
+ the config options have their corresponding
+ variable.
+
+ ASTERISK-29847 #close
+
+
+#### bundled_pjproject: Make it easier to hack
+ Author: George Joseph
+ Date: 2021-11-30
+
+ There are times when you need to troubleshoot issues with bundled
+ pjproject or add new features that need to be pushed upstream
+ but...
+
+ * The source directory created by extracting the pjproject tarball
+ is not scanned for code changes so you have to keep forcing
+ rebuilds.
+ * The source directory isn't a git repo so you can't easily create
+ patches, do git bisects, etc.
+ * Accidentally doing a make distclean will ruin your day by wiping
+ out the source directory, and your changes.
+ * etc.
+
+ This commit makes that easier.
+ See third-party/pjproject/README-hacking.md for the details.
+
+ ASTERISK-29824
+
+
+#### utils.c: Remove all usages of ast_gethostbyname()
+ Author: Sean Bright
+ Date: 2021-12-24
+
+ gethostbyname() and gethostbyname_r() are deprecated in favor of
+ getaddrinfo() which we use in the ast_sockaddr family of functions.
+
+ ASTERISK-29819 #close
+
+
+#### say.conf: fix 12pm noon logic
+ Author: Naveen Albert
+ Date: 2021-12-13
+
+ Fixes 12pm noon incorrectly returning 0/a.m.
+ Also fixes a misspelling typo in the config.
+
+ ASTERISK-29695 #close
+
+
+#### pjproject: Fix incorrect unescaping of tokens during parsing
+ Author: Sean Bright
+ Date: 2022-01-04
+
+ ASTERISK-29664 #close
+
+
+#### app_queue.c: Support for Nordic syntax in announcements
+ Author: Mark Petersen
+ Date: 2021-12-30
+
+ adding support for playing the correct en/et for nordic languages
+ by adding 'n' for neuter gender in the relevant ast_say_number
+
+ ASTERISK-29827
+
+
+#### dsp: Add define macro for DTMF_MATRIX_SIZE
+ Author: Naveen Albert
+ Date: 2021-12-23
+
+ Adds the macro DTMF_MATRIX_SIZE to replace
+ the magic number 4 sprinkled throughout
+ dsp.c.
+
+ ASTERISK-29815 #close
+
+
+#### ami: Add AMI event for Wink
+ Author: Naveen Albert
+ Date: 2022-01-03
+
+ Adds an AMI event for a wink frame.
+
+ ASTERISK-29830 #close
+
+
+#### cli: Add module refresh command
+ Author: Naveen Albert
+ Date: 2021-12-15
+
+ Adds a command to the CLI to unload and then
+ load a module. This makes it easier to perform
+ these operations which are often done
+ subsequently to load a new version of a module.
+
+ "module reload" already refers to reloading of
+ configuration, so the name "refresh" is chosen
+ instead.
+
+ ASTERISK-29807 #close
+
+
+#### app_mp3: Throw warning on nonexistent stream
+ Author: Naveen Albert
+ Date: 2022-01-03
+
+ Currently, the MP3Player application doesn't
+ emit a warning if attempting to play a stream
+ which no longer exists. This can be a common
+ scenario as many mp3 streams are valid at some
+ point but can disappear at any time.
+
+ Now a warning is thrown if attempting to play
+ a nonexistent MP3 stream, instead of silently
+ exiting.
+
+ ASTERISK-29829 #close
+
+
+#### documentation: Add missing AMI documentation
+ Author: Naveen Albert
+ Date: 2021-12-13
+
+ Adds missing documentation for some channel,
+ bridge, and queue events.
+
+ ASTERISK-24427
+ ASTERISK-29515
+
+
+#### tcptls.c: refactor client connection to be more robust
+ Author: Kevin Harwell
+ Date: 2021-11-15
+
+ The current TCP client connect code, blocks and does not handle EINTR
+ error case.
+
+ This patch makes the client socket non-blocking while connecting,
+ ensures a connect does not immediately fail due to EINTR "errors",
+ and adds a connect timeout option.
+
+ The original client start call sets the new timeout option to
+ "infinite", thus making sure old, orginal behavior is retained.
+
+ ASTERISK-29746 #close
+
+
+#### app_sf: Add full tech-agnostic SF support
+ Author: Naveen Albert
+ Date: 2021-12-13
+
+ Adds tech-agnostic support for SF signaling
+ by adding SF sender and receiver applications
+ as well as Dial integration.
+
+ ASTERISK-29802 #close
+
+
+#### app_queue: Fix hint updates, allow dup. hints
+ Author: Steve Davies
+ Date: 2021-12-15
+
+ A previous patch for ASTERISK_29578 caused a 'leak' of
+ extension state information across queues, causing the
+ state of the first member of unrelated queues to be
+ updated in addition to the correct member. Which queues
+ and members depended on the order of queues in the
+ iterator.
+
+ Additionally, it is possible to use the same 'hint:' on
+ multiple queue members, so the update cannot break out
+ of the update loop early when a match is found.
+
+ ASTERISK-29806 #close
+
+
+#### say.c: Honor requests for DTMF interruption.
+ Author: Sean Bright
+ Date: 2021-12-23
+
+ SayAlpha, SayAlphaCase, SayDigits, SayMoney, SayNumber, SayOrdinal,
+ and SayPhonetic all claim to allow DTMF interruption if the
+ SAY_DTMF_INTERRUPT channel variable is set to a truthy value, but we
+ are failing to break out of a given 'say' application if DTMF actually
+ occurs.
+
+ ASTERISK-29816 #close
+
+
+#### res_pjsip_sdp_rtp: Preserve order of RTP codecs
+ Author: Florentin Mayer
+ Date: 2021-11-16
+
+ The ast_rtp_codecs_payloads functions do not preserve the order in which
+ the payloads were specified on an incoming SDP media line. This leads to
+ a problem with the codec negotiation functionality, as the format
+ capabilities of the stream are extracted from the ast_rtp_codecs. This
+ commit moves the ast_rtp_codec to ast_format conversion to the place
+ where the order is still known.
+
+ ASTERISK-28863
+ ASTERISK-29320
+
+
+#### bridge: Unlock channel during Local peer check.
+ Author: Joshua C. Colp
+ Date: 2021-12-27
+
+ It's not safe to keep the channel locked while locking
+ the peer Local channel, as it can result in a deadlock.
+
+ This change unlocks it during this time but keeps the
+ bridge locked to ensure nothing changes about the bridge.
+
+ ASTERISK-29821
+
+
+#### test_time.c: Tolerate DST transitions
+ Author: Josh Soref
+ Date: 2021-11-07
+
+ When test_timezone_watch runs very near a DST transition,
+ two time zones that would otherwise be expected to report the same
+ time can differ because of the DST transition.
+
+ Instead of having the test fail when this happens, report the
+ times, time zones, and dst flags.
+
+ ASTERISK-29722
+
+
+#### bundled_pjproject: Add more support for multipart bodies
+ Author: George Joseph
+ Date: 2021-12-14
+
+ Adding upstream patch for pull request...
+ https://github.com/pjsip/pjproject/pull/2920
+ ---------------------------------------------------------------
+
+ sip_inv: Additional multipart support (#2919)
+
+ sip_inv.c:inv_check_sdp_in_incoming_msg() deals with multipart
+ message bodies in rdata correctly. In the case where early media is
+ involved though, the existing sdp has to be retrieved from the last
+ tdata sent in this transaction. This, however, always assumes that
+ the sdp sent is in a non-multipart body. While there's a function
+ to retrieve the sdp from multipart and non-multpart rdata bodies,
+ no similar function for tdata exists. So...
+
+ * The existing pjsip_rdata_get_sdp_info2 was refactored to
+ find the sdp in any body, multipart or non-multipart, and
+ from either an rdata or tdata. The new function is
+ pjsip_get_sdp_info. This new function detects whether the
+ pjsip_msg->body->data is the text representation of the sdp
+ from an rdata or an existing pjmedia_sdp_session object
+ from a tdata, or whether pjsip_msg->body is a multipart
+ body containing either of the two sdp formats.
+
+ * The exsting pjsip_rdata_get_sdp_info and pjsip_rdata_get_sdp_info2
+ functions are now wrappers that get the body and Content-Type
+ header from the rdata and call pjsip_get_sdp_info.
+
+ * Two new wrappers named pjsip_tdata_get_sdp_info and
+ pjsip_tdata_get_sdp_info2 have been created that get the body
+ from the tdata and call pjsip_get_sdp_info.
+
+ * inv_offer_answer_test.c was updated to test multipart scenarios.
+
+ ASTERISK-29804
+
+
+#### ast_coredumper: Fix deleting results when output dir is set
+ Author: Frederic Van Espen
+ Date: 2021-12-09
+
+ When OUTPUTDIR is set to another directory and the
+ --delete-results-after is set, the resulting txt files are
+ not deleted.
+
+ ASTERISK-29794 #close
+
+
+#### pbx_variables: initialize uninitialized variable
+ Author: Naveen Albert
+ Date: 2021-12-13
+
+ The variable cp4 in a variable substitution function
+ can potentially be used without being initialized
+ currently. This causes Asterisk to no longer compile.
+
+ This initializes cp4 to NULL to make the compiler
+ happy.
+
+ ASTERISK-29803 #close
+
+
+#### app_queue.c: added DIALEDPEERNUMBER on outgoing channel
+ Author: Mark Petersen
+ Date: 2021-12-08
+
+ added that we set DIALEDPEERNUMBER on the outgoing channels
+ so it is avalible in b(content^extension^line)
+ this add the same behaviour as Dial
+
+ ASTERISK-29795
+
+
+#### http.c: Add ability to create multiple HTTP servers
+ Author: Kevin Harwell
+ Date: 2021-11-15
+
+ Previously, it was only possible to have one HTTP server in Asterisk.
+ With this patch it is now possible to have multiple HTTP servers
+ listening on different addresses.
+
+ Note, this behavior has only been made available through an API call
+ from within the TEST_FRAMEWORK. Specifically, this feature has been
+ added in order to allow unit test to create/start and stop servers,
+ if one has not been enabled through configuration.
+
+
+#### app.c: Throw warnings for nonexistent options
+ Author: Naveen Albert
+ Date: 2021-12-13
+
+ Currently, Asterisk doesn't throw warnings if options
+ are passed into applications that don't accept them.
+ This can confuse users if they're unaware that they
+ are doing something wrong.
+
+ This adds an additional check to parse_options so that
+ a warning is thrown anytime an option is parsed that
+ doesn't exist in the parsing application, so that users
+ are notified of the invalid usage.
+
+ ASTERISK-29801 #close
+
+
+#### app_voicemail.c: Support for Danish syntax in VM
+ Author: Mark Petersen
+ Date: 2021-12-08
+
+ added support for playing the correct plural sound file
+ dependen on where you have 1 or multipe messages
+ based on the existing SE/NO code
+
+ ASTERISK-29797
+
+
+#### app_sendtext: Add ReceiveText application
+ Author: Naveen Albert
+ Date: 2021-11-17
+
+ Adds a ReceiveText application that can be used in
+ conjunction with SendText. Currently, there is no
+ way in Asterisk to receive text in the dialplan
+ (or anywhere else, really). This allows for Asterisk
+ to be the recipient of text instead of just the sender.
+
+ ASTERISK-29759 #close
+
+
+#### strings: Fix enum names in comment examples
+ Author: Naveen Albert
+ Date: 2021-12-12
+
+ The enum values for ast_strsep_flags includes
+ AST_STRSEP_STRIP. However, some comments reference
+ AST_SEP_STRIP, which doesn't exist. This fixes
+ these comments to use the correct value.
+
+ ASTERISK-29800 #close
+
+
+#### pbx_variables: Increase parsing capabilities of MSet
+ Author: Naveen Albert
+ Date: 2021-11-20
+
+ Currently MSet can only parse a maximum of 24 variables.
+ If more variables are provided to MSet, the 24th variable
+ will simply contain the remainder of the string and the
+ remaining variables thereafter will never get set.
+
+ This increases the number of variables that can be parsed
+ in one go from 24 to 99. Additionally, documentation is added
+ since this limitation is currently undocumented and is
+ confusing to users who encounter this limitation.
+
+ ASTERISK-29766 #close
+
+
+#### chan_sip: Fix crash when accessing RURI before initiating outgoing call
+ Author: Naveen Albert
+ Date: 2021-11-24
+
+ Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before
+ initiating an outgoing call will cause Asterisk to crash. This is
+ because a null field is accessed, resulting in an offset from null and
+ subsequent memory access violation.
+
+ Since RURI is not guaranteed to exist, we now check if the base
+ pointer is non-null before calculating an offset.
+
+ ASTERISK-29772
+
+
+#### func_json: Adds JSON_DECODE function
+ Author: Naveen Albert
+ Date: 2021-10-25
+
+ Adds the JSON_DECODE function for parsing JSON in the
+ dialplan. JSON parsing already exists in the Asterisk
+ core and is used for many different things. This
+ function exposes the basic parsing capability to
+ the user in the dialplan, for instance, in conjunction
+ with CURL for using API responses.
+
+ ASTERISK-29706 #close
+
+
+#### configs: Updates to sample configs
+ Author: Naveen Albert
+ Date: 2021-11-17
+
+ Includes some minor updates to extensions.conf
+ and iax.conf. In particular, the demonstration
+ of macros in extensions.conf is removed, as
+ Macro is deprecated and will be removed soon.
+ These examples have been replaced with examples
+ demonstrating the usage of Gosub instead.
+
+ The older exten => ...,n syntax is also mostly
+ replaced with the same keyword to demonstrate the
+ newer, more concise way of defining extensions.
+
+ IAXTEL no longer exists, so this example is replaced
+ with something more generic.
+
+ Some documentation is also added to extensions.conf
+ and iax.conf to clarify some of the new expanded
+ encryption capabilities with IAX2.
+
+ ASTERISK-29758 #close
+
+
+#### pbx: Add variable substitution API for extensions
+ Author: Naveen Albert
+ Date: 2021-11-15
+
+ Currently, variable substitution involving dialplan
+ extensions is quite clunky since it entails obtaining
+ the current dialplan location, backing it up, storing
+ the desired variables for substitution on the channel,
+ performing substitution, then restoring the original
+ location.
+
+ In addition to being clunky, things could also go wrong
+ if an async goto were to occur and change the dialplan
+ location during a substitution.
+
+ Fundamentally, there's no reason it needs to be done this
+ way, so new API is added to allow for directly passing in
+ the dialplan location for the purposes of variable
+ substitution so we don't need to mess with the channel
+ information anymore. Existing API is not changed.
+
+ ASTERISK-29745 #close
+
+
+#### CHANGES: Correct reference to configuration file.
+ Author: Sean Bright
+ Date: 2021-12-11
+
+
+#### app_mf: Add full tech-agnostic MF support
+ Author: Naveen Albert
+ Date: 2021-09-22
+
+ Adds tech-agnostic support for MF signaling by adding
+ MF sender and receiver applications as well as Dial
+ integration.
+
+ ASTERISK-29496-mf #do-not-close
+
+
+#### xmldoc: Avoid whitespace around value for parameter/required.
+ Author: Alexander Traud
+ Date: 2021-12-06
+
+ Otherwise, the value 'false' was not found in the enumerated set of
+ the XML DTD for the XML attribute 'required' in the XML element
+ 'parameter'. Therefore, DTD validation of the runtime XML failed.
+
+ ASTERISK-29790
+
+
+#### progdocs: Fix Doxygen left-overs.
+ Author: Alexander Traud
+ Date: 2021-12-04
+
+
+#### xmldoc: Correct definition for XML element 'matchInfo'.
+ Author: Alexander Traud
+ Date: 2021-12-06
+
+ ASTERISK-29791
+
+
+#### progdocs: Update Makefile.
+ Author: Alexander Traud
+ Date: 2021-11-23
+
+ In developer mode, use internal documentation as well.
+ This should produce no warnings. Fix yours!
+
+ In noisy mode, output all possible warnings of Doxygen.
+ This creates zillion of warnings. Double-check your current module!
+
+ Any warnings are in the file './doxygen.log'. Beside that, this change
+ avoids deprecated parameters because the configuration file for Doxygen
+ contains only those parameters which differ from the default. This
+ avoids the need to update the file on each run. Furthermore, it adds
+ AST_VECTOR to be expanded. Finally, the default name for that file is
+ Doxyfile. Therefore, let us use that!
+
+ ASTERISK-26991
+ ASTERISK-20259
+
+
+#### res_pjsip_sdp_rtp: Do not warn on unknown sRTP crypto suites.
+ Author: Alexander Traud
+ Date: 2021-12-03
+
+ res_sdp_crypto_parse_offer(.) emits many log messages already.
+
+ ASTERISK-29785
+
+
+#### channel: Short-circuit ast_channel_get_by_name() on empty arg.
+ Author: Sean Bright
+ Date: 2021-11-30
+
+ We know that passing a NULL or empty argument to
+ ast_channel_get_by_name() will never result in a matching channel and
+ will always result in an error being emitted, so just short-circuit
+ out in that case.
+
+ ASTERISK-28219 #close
+
+
+#### res_rtp_asterisk: Addressing possible rtp range issues
+ Author: Mike Bradeen
+ Date: 2021-10-26
+
+ res/res_rtp_asterisk.c: Adding 1 to rtpstart if it is deteremined
+ that rtpstart was configured to be an odd value. Also adding a loop
+ counter to prevent a possible infinite loop when looking for a free
+ port.
+
+ ASTERISK-27406
+
+
+#### apps/app_dial.c: HANGUPCAUSE reason code for CANCEL is set to AST_CAUSE_NORMAL..
+ Author: Mark Petersen
+ Date: 2021-08-24
+
+ changed that when we recive a CANCEL that we set HANGUPCAUSE to AST_CAUSE_NORMAL_CLEARING
+
+ ASTERISK-28053
+ Reported by: roadkill
+
+
+#### res: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-19
+
+ These are the remaining issues found in /res.
+
+ ASTERISK-29761
+
+
+#### res_fax_spandsp: Add spandsp 3.0.0+ compatibility
+ Author: Dustin Marquess
+ Date: 2021-11-08
+
+ Newer versions of spandsp did refactoring of code to add new features
+ like color FAXing. This refactoring broke backwards compatibility.
+ Add support for the new version while retaining support for 0.0.6.
+
+ ASTERISK-29729 #close
+
+
+#### main: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-19
+
+ ASTERISK-29763
+
+
+#### progdocs: Fix for Doxygen, the hidden parts.
+ Author: Alexander Traud
+ Date: 2021-11-27
+
+ ASTERISK-29779
+
+
+#### progdocs: Fix grouping for latest Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-12
+
+ Since Doxygen 1.8.16, a special comment block is required. Otherwise
+ (pure C comment), the group command is ignored. Additionally, several
+ unbalanced group commands were fixed.
+
+ ASTERISK-29732
+
+
+#### documentation: Standardize examples
+ Author: Naveen Albert
+ Date: 2021-11-25
+
+ Most examples in the XML documentation use the
+ example tag to demonstrate examples, which gets
+ parsed specially in the Wiki to make it easier
+ to follow for users.
+
+ This fixes a few modules to use the example
+ tag instead of vanilla para tags to bring them
+ in line with the standard syntax.
+
+ ASTERISK-29777 #close
+
+
+#### config.c: Prevent UB in ast_realtime_require_field.
+ Author: Sean Bright
+ Date: 2021-11-28
+
+ A backend's implementation of the realtime 'require' function may call
+ va_arg() and then fail, leaving the va_list in an undefined
+ state. Pass a copy of the va_list instead.
+
+ ASTERISK-29771 #close
+
+
+#### app_voicemail: Refactor email generation functions
+ Author: Naveen Albert
+ Date: 2021-11-01
+
+ Refactors generic functions used for email generation
+ into utils.c so that they can be used by multiple
+ modules, including app_voicemail and app_minivm,
+ to avoid code duplication.
+
+ ASTERISK-29715 #close
+
+
+#### stir/shaken: Avoid a compiler extension of GCC.
+ Author: Alexander Traud
+ Date: 2021-11-25
+
+ ASTERISK-29776
+
+
+#### progdocs: Remove outdated references in doxyref.h.
+ Author: Alexander Traud
+ Date: 2021-11-23
+
+ ASTERISK-29773
+
+
+#### logger: use __FUNCTION__ instead of __PRETTY_FUNCTION__
+ Author: Jaco Kroon
+ Date: 2021-10-28
+
+ This avoids a few long-name overflows, at the cost of less instructive
+ names in the case of C++ (specifically overloaded functions and class
+ methods). This in turn is offset against the fact that we're logging
+ the filename and line numbers in any case.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### xmldoc: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-20
+
+ ASTERISK-29765
+
+
+#### astobj2.c: Fix core when ref_log enabled
+ Author: Mike Bradeen
+ Date: 2021-11-16
+
+ In the AO2_ALLOC_OPT_LOCK_NOLOCK case the referenced obj
+ structure is freed, but is then referenced later if ref_log is
+ enabled. The change is to store the obj->priv_data.options value
+ locally and reference it instead of the value from the freed obj
+
+ ASTERISK-29730
+
+
+#### channels: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-19
+
+ ASTERISK-29762
+
+
+#### bridge: Deny full Local channel pair in bridge.
+ Author: Joshua C. Colp
+ Date: 2021-11-16
+
+ Local channels are made up of two pairs - the 1 and 2
+ sides. When a frame goes in one side, it comes out the
+ other. Back and forth. When both halves are in a
+ bridge this creates an infinite loop of frames.
+
+ This change makes it so that bridging no longer
+ allows both of these sides to exist in the same
+ bridge.
+
+ ASTERISK-29748
+
+
+#### res_tonedetect: Add call progress tone detection
+ Author: Naveen Albert
+ Date: 2021-11-06
+
+ Makes basic call progress tone detection available
+ in a tech-agnostic manner with the addition of the
+ ToneScan application. This can determine if the channel
+ has encountered a busy signal, SIT tones, dial tone,
+ modem, fax machine, etc. A few basic async progress
+ tone detect options are also added to the TONE_DETECT
+ function.
+
+ ASTERISK-29720 #close
+
+
+#### rtp_engine: Add type field for JSON RTCP Report stasis messages
+ Author: Boris P. Korzun
+ Date: 2021-11-08
+
+ ASTERISK-29727 #close
+
+
+#### odbc: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ ASTERISK-29754
+
+
+#### parking: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ ASTERISK-29753
+
+
+#### res_ari: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ ASTERISK-29756
+
+
+#### frame: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ ASTERISK-29755
+
+
+#### ari-stubs: Avoid 'is' as comparism with an literal.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ Python 3.9.7 gave a syntax warning.
+
+
+#### BuildSystem: Consistently allow 'ye' even for Jansson.
+ Author: Alexander Traud
+ Date: 2021-11-08
+
+ Furthermore, consistently use not 'No' but ':' for non-existent file
+ paths. Finally, use the same pattern for checking file paths:
+ a) = ":"
+ b) != "x:"
+
+
+#### stasis: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-16
+
+ ASTERISK-29750
+
+
+#### app: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-17
+
+ ASTERISK-29752
+
+
+#### res_xmpp: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-16
+
+ ASTERISK-29749
+
+
+#### channel: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-16
+
+ ASTERISK-29751
+
+
+#### chan_iax2: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-13
+
+ ASTERISK-29737
+
+
+#### res_pjsip: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-16
+
+ ASTERISK-29747
+
+
+#### bridges: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-15
+
+ ASTERISK-29743
+
+
+#### addons: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-15
+
+ ASTERISK-29742
+
+
+#### apps: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-15
+
+ ASTERISK-29740
+
+
+#### tests: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-15
+
+ ASTERISK-29741
+
+
+#### progdocs: Avoid multiple use of section labels.
+ Author: Alexander Traud
+ Date: 2021-11-12
+
+ ASTERISK-29735
+
+
+#### progdocs: Use Doxygen \example correctly.
+ Author: Alexander Traud
+ Date: 2021-11-12
+
+ ASTERISK-29734
+
+
+#### bridge_channel: Fix for Doxygen.
+ Author: Alexander Traud
+ Date: 2021-11-13
+
+ ASTERISK-29736
+
+
+#### progdocs: Avoid 'name' with Doxygen \file.
+ Author: Alexander Traud
+ Date: 2021-11-12
+
+ Fixes four misuses of the parameter 'name'. Additionally, for
+ consistency and to avoid such an issue in future, those few other
+ places, which used '\file name', were changed just to '\file'. Then,
+ Doxygen uses the name of the current file.
+
+ ASTERISK-29733
+
+
+#### app_morsecode: Fix deadlock
+ Author: Naveen Albert
+ Date: 2021-11-15
+
+ Fixes a deadlock in app_morsecode caused by locking
+ the channel twice when reading variables from the
+ channel. The duplicate lock is simply removed.
+
+ ASTERISK-29744 #close
+
+
+#### app_read: Fix custom terminator functionality regression
+ Author: Naveen Albert
+ Date: 2021-10-25
+
+ Currently, when the t option is specified with no arguments,
+ the # character is still treated as a terminator, even though
+ no character should be treated as a terminator.
+
+ This is because a previous regression fix was modified to
+ remove the use of NULL as a default altogether. However,
+ NULL and an empty string actually refer to different
+ arrangements and should be treated differently. NULL is the
+ default terminator (#), while an empty string removes the
+ terminator altogether. This is the behavior being used by
+ the rest of the core.
+
+ Additionally, since S_OR catches empty strings as well as
+ NULL (not intended), this is changed to a ternary operator
+ instead, which fixes the behavior.
+
+ ASTERISK-29705 #close
+
+
+#### res_pjsip_callerid: Fix OLI parsing
+ Author: Naveen Albert
+ Date: 2021-10-24
+
+ Fix parsing of ANI2/OLI information, since it was previously
+ parsing the user, when it should have been parsing other_param.
+
+ Also improves the parsing by using pjproject native functions
+ rather than trying to parse the parameters ourselves like
+ chan_sip did. A previous attempt at this caused a crash, but
+ this works correctly now.
+
+ ASTERISK-29703 #close
+
+
+#### build_tools: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ binutils
+
+ ASTERISK-29714
+
+
+#### contrib: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ standard
+ increase
+ comments
+ valgrind
+ promiscuous
+ editing
+ libtonezone
+ storage
+ aggressive
+ whitespace
+ russellbryant
+ consecutive
+ peternixon
+
+ ASTERISK-29714
+
+
+#### codecs: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ voiced
+ denumerator
+ codeword
+ upsampling
+ constructed
+ residual
+ subroutine
+ conditional
+ quantizing
+ courtesy
+ number
+
+ ASTERISK-29714
+
+
+#### formats: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ truncate
+
+ ASTERISK-29714
+
+
+#### CREDITS: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ contributors
+
+ ASTERISK-29714
+
+
+#### addons: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ definition
+ listener
+ fastcopy
+ logical
+ registration
+ classify
+ documentation
+ explicitly
+ dialed
+ endpoint
+ elements
+ arithmetic
+ might
+ prepend
+ byte
+ terminal
+ inquiry
+ skipping
+ aliases
+ calling
+ absent
+ authentication
+ transmit
+ their
+ ericsson
+ disconnecting
+ redir
+ items
+ client
+ adapter
+ transmitter
+ existing
+ satisfies
+ pointer
+ interval
+ supplied
+
+ ASTERISK-29714
+
+
+#### configs: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ password
+ excludes
+ undesirable
+ checksums
+ through
+ screening
+ interpreting
+ database
+ causes
+ initiation
+ member
+ busydetect
+ defined
+ severely
+ throughput
+ recognized
+ counter
+ require
+ indefinitely
+ accounts
+
+ ASTERISK-29714
+
+
+#### doc: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ transparent
+ roughly
+
+ ASTERISK-29714
+
+
+#### menuselect: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ dependency
+ unless
+ random
+ dependencies
+ delimited
+ randomly
+ modules
+
+ ASTERISK-29714
+
+
+#### include: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ activities
+ forward
+ occurs
+ unprepared
+ association
+ compress
+ extracted
+ doubly
+ callback
+ prometheus
+ underlying
+ keyframe
+ continue
+ convenience
+ calculates
+ ignorepattern
+ determine
+ subscribers
+ subsystem
+ synthetic
+ applies
+ example
+ manager
+ established
+ result
+ microseconds
+ occurrences
+ unsuccessful
+ accommodates
+ related
+ signifying
+ unsubscribe
+ greater
+ fastforward
+ itself
+ unregistering
+ using
+ translator
+ sorcery
+ implementation
+ serializers
+ asynchronous
+ unknowingly
+ initialization
+ determining
+ category
+ these
+ persistent
+ propagate
+ outputted
+ string
+ allocated
+ decremented
+ second
+ cacheability
+ destructor
+ impaired
+ decrypted
+ relies
+ signaling
+ based
+ suspended
+ retrieved
+ functions
+ search
+ auth
+ considered
+
+ ASTERISK-29714
+
+
+#### UPGRADE.txt: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ themselves
+ support
+ received
+
+ ASTERISK-29714
+
+
+#### bridges: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ multiplication
+ potentially
+ iteration
+ interaction
+ virtual
+ synthesis
+ convolve
+ initializes
+ overlap
+
+ ASTERISK-29714
+
+
+#### apps: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ simultaneously
+ administrator
+ directforward
+ attachfmt
+ dailplan
+ automatically
+ applicable
+ nouns
+ explicit
+ outside
+ sponsored
+ attachment
+ audio
+ spied
+ doesn't
+ counting
+ encoded
+ implements
+ recursively
+ emailaddress
+ arguments
+ queuerules
+ members
+ priority
+ output
+ advanced
+ silencethreshold
+ brazilian
+ debugging
+ argument
+ meadmin
+ formatting
+ integrated
+ sneakiness
+
+ ASTERISK-29714
+
+
+#### channels: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ appease
+ permanently
+ overriding
+ residue
+ silliness
+ extension
+ channels
+ globally
+ reference
+ japanese
+ group
+ coordinate
+ registry
+ information
+ inconvenience
+ attempts
+ cadence
+ payloads
+ presence
+ provisioning
+ mimics
+ behavior
+ width
+ natively
+ syslabel
+ not owning
+ unquelch
+ mostly
+ constants
+ interesting
+ active
+ unequipped
+ brodmann
+ commanding
+ backlogged
+ without
+ bitstream
+ firmware
+ maintain
+ exclusive
+ practically
+ structs
+ appearance
+ range
+ retransmission
+ indication
+ provisional
+ associating
+ always
+ whether
+ cyrillic
+ distinctive
+ components
+ reinitialized
+ initialized
+ capability
+ switches
+ occurring
+ happened
+ outbound
+
+ ASTERISK-29714
+
+
+#### tests: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ mounting
+ jitterbuffer
+ thrashing
+ original
+ manipulating
+ entries
+ actual
+ possibility
+ tasks
+ options
+ positives
+ taskprocessor
+ other
+ dynamic
+ declarative
+
+ ASTERISK-29714
+
+
+#### CHANGES: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ issuing
+ execution
+ bridging
+ alert
+ respective
+ unlikely
+ confbridge
+ offered
+ negotiation
+ announced
+ engineer
+ systems
+ inherited
+ passthrough
+ functionality
+ supporting
+ conflicts
+ semantically
+ monitor
+ specify
+ specifiable
+
+ ASTERISK-29714
+
+
+#### funcs: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ effectively
+ emitted
+ expect
+ anthony
+
+ ASTERISK-29714
+
+
+#### pbx: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ process
+ populate
+ with
+ africa
+ accessing
+ contexts
+ exercise
+ university
+ organizations
+ withhold
+ maintaining
+ independent
+ rotation
+ ignore
+ eventname
+
+ ASTERISK-29714
+
+
+#### main: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ analysis
+ nuisance
+ converting
+ although
+ transaction
+ desctitle
+ acquire
+ update
+ evaluate
+ thousand
+ this
+ dissolved
+ management
+ integrity
+ reconstructed
+ decrement
+ further on
+ irrelevant
+ currently
+ constancy
+ anyway
+ unconstrained
+ featuregroups
+ right
+ larger
+ evaluated
+ encumbered
+ languages
+ digits
+ authoritative
+ framing
+ blindxfer
+ tolerate
+ traverser
+ exclamation
+ perform
+ permissions
+ rearrangement
+ performing
+ processing
+ declension
+ happily
+ duplicate
+ compound
+ hundred
+ returns
+ elicit
+ allocate
+ actually
+ paths
+ inheritance
+ atxferdropcall
+ earlier
+ synchronization
+ multiplier
+ acknowledge
+ across
+ against
+ thousands
+ joyous
+ manipulators
+ guaranteed
+ emulating
+ soundfile
+
+ ASTERISK-29714
+
+
+#### utils: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ command-line
+ immediately
+ extensions
+ momentarily
+ mustn't
+ numbered
+ bytes
+ caching
+
+ ASTERISK-29714
+
+
+#### Makefile: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ libraries
+ install
+ overwrite
+
+ ASTERISK-29714
+
+
+#### res: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ identifying
+ structures
+ actcount
+ initializer
+ attributes
+ statement
+ enough
+ locking
+ declaration
+ userevent
+ provides
+ unregister
+ session
+ execute
+ searches
+ verification
+ suppressed
+ prepared
+ passwords
+ recipients
+ event
+ because
+ brief
+ unidentified
+ redundancy
+ character
+ the
+ module
+ reload
+ operation
+ backslashes
+ accurate
+ incorrect
+ collision
+ initializing
+ instance
+ interpreted
+ buddies
+ omitted
+ manually
+ requires
+ queries
+ generator
+ scheduler
+ configuration has
+ owner
+ resource
+ performed
+ masquerade
+ apparently
+ routable
+
+ ASTERISK-29714
+
+
+#### rest-api-templates: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ overwritten
+ descendants
+
+ ASTERISK-29714
+
+
+#### agi: Spelling fixes
+ Author: Josh Soref
+ Date: 2021-10-30
+
+ Correct typos of the following word families:
+
+ pretend
+ speech
+
+ ASTERISK-29714
+
+
+#### CI: Rename 'master' node to 'built-in'
+ Author: George Joseph
+ Date: 2021-11-08
+
+ Jenkins renamed the 'master' node to 'built-in' in version
+ 2.319 so we have to adjust as well.
+
+
+#### BuildSystem: In POSIX sh, == in place of = is undefined.
+ Author: Alexander Traud
+ Date: 2021-11-08
+
+ ASTERISK-29724
+
+
+#### pbx.c: Don't remove dashes from hints on reload.
+ Author: Sean Bright
+ Date: 2021-11-08
+
+ When reloading dialplan, hints created dynamically would lose any dash
+ characters. Now we ignore those dashes if we are dealing with a hint
+ during a reload.
+
+ ASTERISK-28040 #close
+
+
+#### sig_analog: Fix truncated buffer copy
+ Author: Naveen Albert
+ Date: 2021-10-24
+
+ Fixes compiler warning caused by a truncated copy of the ANI2 into a
+ buffer of size 10. This could prevent the null terminator from being
+ copied if the copy value exceeds the size of the buffer. This increases
+ the buffer size to 101 to ensure there is no way for truncation to occur.
+
+ ASTERISK-29702 #close
+
+
+#### app_voicemail: Fix phantom voicemail bug on rerecord
+ Author: Naveen Albert
+ Date: 2021-10-24
+
+ If users are able to press # for options while leaving
+ a message and then press 3 to rerecord the message, if
+ the caller hangs up during the rerecord prompt but before
+ Asterisk starts recording a message, then an "empty"
+ voicemail gets processed whereby an email gets sent out
+ notifying the user of a 0:00 duration message. The file
+ doesn't actually exist, so playback will fail since there
+ was no message to begin with.
+
+ This adds a check after the streaming of the rerecord
+ announcement to see if the caller has hung up. If so,
+ we bail out early so that we can clean up properly.
+
+ ASTERISK-29391 #close
+
+
+#### chan_iax2: Allow both secret and outkey at dial time
+ Author: Naveen Albert
+ Date: 2021-10-26
+
+ Historically, the dial syntax for IAX2 has held that
+ an outkey (used only for RSA authenticated calls)
+ and a secret (used only for plain text and MD5 authenticated
+ calls, historically) were mutually exclusive, and thus
+ the same position in the dial string was used for both
+ values.
+
+ Now that encryption is possible with RSA authentication,
+ this poses a limitation, since encryption requires a
+ secret and RSA authentication requires an outkey. Thus,
+ the dial syntax is extended so that both a secret and
+ an outkey can be specified.
+
+ The new extended syntax is backwards compatible with the
+ old syntax. However, a secret can now be specified after
+ the outkey, or the outkey can be specified after the secret.
+ This makes it possible to spawn an encrypted RSA authenticated
+ call without a corresponding peer being predefined in iax.conf.
+
+ ASTERISK-29707 #close
+
+
+#### res_snmp: As build tool, prefer pkg-config over net-snmp-config.
+ Author: Alexander Traud
+ Date: 2021-10-28
+
+ ASTERISK-29709
+
+
+#### res_config_sqlite: Remove deprecated module.
+ Author: Alexander Traud
+ Date: 2021-11-04
+
+ ASTERISK-29717
+
+
+#### stasis: Avoid 'dispatched' as unused variable in normal mode.
+ Author: Alexander Traud
+ Date: 2021-10-28
+
+ ASTERISK-29710
+
+
+#### various: Fix GCC 11.2 compilation issues.
+ Author: Sean Bright
+ Date: 2021-10-29
+
+ * Initialize some variables that are never used anyway.
+
+ * Use valid pointers instead of integers cast to void pointers when
+ calling pthread_setspecific().
+
+ ASTERISK-29711 #close
+ ASTERISK-29713 #close
+
+
+#### ast_coredumper: Refactor to better find things
+ Author: George Joseph
+ Date: 2021-09-09
+
+ The search for a running asterisk when --running is used
+ has been greatly simplified and in the event it doesn't
+ work, you can now specify a pid to use on the command
+ line with --pid.
+
+ The search for asterisk modules when --tarball-coredumps
+ is used has been enhanced to have a better chance of finding
+ them and in the event it doesn't work, you can now specify
+ --libdir on the command line to indicate the library directory
+ where they were installed.
+
+ The DATEFORMAT variable was renamed to DATEOPTS and is now
+ passed to the 'date' utility rather than running DATEFORMAT
+ as a command.
+
+ The coredump and output files are now renamed with DATEOPTS.
+ This can be disabled by specifying --no-rename.
+
+ Several confusing and conflicting options were removed:
+ --append-coredumps
+ --conffile
+ --no-default-search
+ --tarball-uniqueid
+
+ The script was re-structured to make it easier for follow.
+
+
+#### strings/json: Add string delimter match, and object create with vars methods
+ Author: Kevin Harwell
+ Date: 2021-10-21
+
+ Add a function to check if there is an exact match a one string between
+ delimiters in another string.
+
+ Add a function that will create an ast_json object out of a list of
+ Asterisk variables. An excludes string can also optionally be passed
+ in.
+
+ Also, add a macro to make it easier to get object integers.
+
+
+#### STIR/SHAKEN: Option split and response codes.
+ Author: Ben Ford
+ Date: 2021-09-21
+
+ The stir_shaken configuration option now has 4 different choices to pick
+ from: off, attest, verify, and on. Off and on behave the same way they
+ do now. Attest will only perform attestation on the endpoint, and verify
+ will only perform verification on the endpoint.
+
+ Certain responses are required to be sent based on certain conditions
+ for STIR/SHAKEN. For example, if we get a Date header that is outside of
+ the time range that is considered valid, a 403 Stale Date response
+ should be sent. This and several other responses have been added.
+
+
+#### app_queue: Add LoginTime field for member in a queue.
+ Author: Rodrigo Ramírez Norambuena
+ Date: 2021-08-25
+
+ Add a time_t logintime to storage a time when a member is added into a
+ queue.
+
+ Also, includes show this time (in seconds) using a 'queue show' command
+ and the field LoginTime for response for AMI events.
+
+ ASTERISK-18069 #close
+
+
+#### res_speech: Add a type conversion, and new engine unregister methods
+ Author: Kevin Harwell
+ Date: 2021-10-21
+
+ Add a new function that converts a speech results type to a string.
+ Also add another function to unregister an engine, but returns a
+ pointer to the unregistered engine object instead of a success/fail
+ integer.
+
+
+#### various: Fix GCC 11 compilation issues.
+ Author: Mike Bradeen
+ Date: 2021-10-07
+
+ test_voicemail_api: Use empty char* for empty_msg_ids.
+ chan_skinny: Fix size of calledParty to be maximum extension.
+ menuselect: Change Makefile to stop deprecated warnings. Added comments
+ test_linkedlist: 'bogus' variable was manually allocated from a macro
+ and the test fails if this happens but the compiler couldn't 'see' this
+ and returns a warning. memset to all 0's after allocation.
+ chan_ooh323: Fixed various indentation issues that triggered misleading
+ indentation warnings.
+
+ ASTERISK-29682
+ Reported by: George Joseph
+
+
+#### apps/app_playback.c: Add 'mix' option to app_playback
+ Author: Shloime Rosenblum
+ Date: 2021-09-20
+
+ I am adding a mix option that will play by filename and say.conf unlike
+ say option that will only play with say.conf. It
+ will look on the format of the name, if it is like say it play with
+ say.conf if not it will play the file name.
+
+ ASTERISK-29662
+
+
+#### BuildSystem: Check for alternate openssl packages
+ Author: George Joseph
+ Date: 2021-10-19
+
+ OpenSSL is one of those packages that often have alternatives
+ with later versions. For instance, CentOS/EL 7 has an
+ openssl package at version 1.0.2 but there's an openssl11
+ package from the epel repository that has 1.1.1. This gets
+ installed to /usr/include/openssl11 and /usr/lib64/openssl11.
+ Unfortunately, the existing --with-ssl and --with-crypto
+ ./configure options expect to point to a source tree and
+ don't work in this situation. Also unfortunately, the
+ checks in ./configure don't use pkg-config.
+
+ In order to make this work with the existing situation, you'd
+ have to run...
+ ./configure --with-ssl=/usr/lib64/openssl11 \
+ --with-crypto=/usr/lib64/openssl11 \
+ CFLAGS=-I/usr/include/openssl11
+
+ BUT... those options don't get passed down to bundled pjproject
+ so when you run make, you have to include the CFLAGS again
+ which is a big pain.
+
+ Oh... To make matters worse, although you can specify
+ PJPROJECT_CONFIGURE_OPTS on the ./configure command line,
+ they don't get saved so if you do a make clean, which will
+ force a re-configure of bundled pjproject, those options
+ don't get used.
+
+ So...
+
+ * In configure.ac... Since pkg-config is installed by install_prereq
+ anyway, we now use it to check for the system openssl >= 1.1.0.
+ If that works, great. If not, we check for the openssl11
+ package. If that works, great. If not, we fall back to just
+ checking for any openssl. If pkg-config isn't installed for some
+ reason, or --with-ssl=<dir> or --with-crypto=<dir> were specified
+ on the ./configure command line, we fall back to the existing
+ logic that uses AST_EXT_LIB_CHECK().
+
+ * The whole OpenSSL check process has been moved up before
+ THIRD_PARTY_CONFIGURE(), which does the initial pjproject
+ bundled configure, is run. This way the results of the above
+ checks, which may result in new include or library directories,
+ is included.
+
+ * Although not strictly needed for openssl, We now save the value of
+ PJPROJECT_CONFIGURE_OPTS in the makeopts file so it can be used
+ again if a re-configure is triggered.
+
+ ASTERISK-29693
+
+
+#### func_talkdetect.c: Fix logical errors in silence detection.
+ Author: Sean Bright
+ Date: 2021-10-14
+
+ There are 3 separate changes here:
+
+ 1. The documentation erroneously stated that the dsp_talking_threshold
+ argument was a number of milliseconds when it is actually an energy
+ level used by the DSP code to classify talking vs. silence.
+
+ 2. Fixes a copy paste error in the argument handling code.
+
+ 3. Don't erroneously switch to the talking state if we aren't actively
+ handling a frame we've classified as talking.
+
+ Patch inspired by one provided by Moritz Fain (License #6961).
+
+ ASTERISK-27816 #close
+
+
+#### configure: Remove unused OpenSSL SRTP check.
+ Author: Sean Bright
+ Date: 2021-10-11
+
+ Discovered while looking at ASTERISK~29684. Usage was removed in change
+ I3c77c7b00b2ffa2e935632097fa057b9fdf480c0.
+
+
+#### build: prevent binary downloads for non x86 architectures
+ Author: Mike Bradeen
+ Date: 2021-10-12
+
+ download_externals: Add check for i686 and i386 (in addition
+ to the current x86_64) and exit if not one of the three.
+
+ ASTERISK-26497
+
+
+#### main/stun.c: fix crash upon STUN request timeout
+ Author: Sebastien Duthil
+ Date: 2021-10-14
+
+ Some ast_stun_request users do not provide a destination address when
+ sending to a connection-mode socket.
+
+ ASTERISK-29691
+
+
+#### Makefile: Use basename in a POSIX-compliant way.
+ Author: Sean Bright
+ Date: 2021-10-07
+
+ If you aren't using GNU coreutils, chances are that your basename
+ doesn't know about the -s argument. Luckily for us, basename does what
+ we need it do even without the -s argument.
+
+
+#### pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+ Author: Mark Murawski
+ Date: 2021-10-05
+
+ Avoid infinite recursion and crash
+
+
+#### chan_iax2: Add encryption for RSA authentication
+ Author: Naveen Albert
+ Date: 2021-05-24
+
+ Adds support for encryption to RSA-authenticated
+ calls. Also prevents crashes if an RSA IAX2 call
+ is initiated to a switch requiring encryption
+ but no secret is provided.
+
+ ASTERISK-20219
+
+
+#### res_pjsip_t38: bind UDPTL sessions like RTP
+ Author: Matthew Kern
+ Date: 2021-07-19
+
+ In res_pjsip_sdp_rtp, the bind_rtp_to_media_address option and the
+ fallback use of the transport's bind address solve problems sending
+ media on systems that cannot send ipv4 packets on ipv6 sockets, and
+ certain other situations. This change extends both of these behaviors
+ to UDPTL sessions as well in res_pjsip_t38, to fix fax-specific
+ problems on these systems, introducing a new option
+ endpoint/t38_bind_udptl_to_media_address.
+
+ ASTERISK-29402
+
+
+#### app_read: Fix null pointer crash
+ Author: Naveen Albert
+ Date: 2021-09-29
+
+ If the terminator character is not explicitly specified
+ and an indications tone is used for reading a digit,
+ there is no null pointer check so Asterisk crashes.
+ This prevents null usage from occuring.
+
+ ASTERISK-29673 #close
+
+
+#### res_rtp_asterisk: fix memory leak
+ Author: Jean Aunis
+ Date: 2021-09-29
+
+ Add missing reference decrement in rtp_deallocate_transport()
+
+ ASTERISK-29671
+
+
+#### main/say.c: Support future dates with Q and q format params
+ Author: Shloime Rosenblum
+ Date: 2021-09-19
+
+ The current versions do not support future dates in all say application when using the 'Q' or 'q' format parameter and says "today" for everything that is greater than today
+
+ ASTERISK-29637
+
+
+#### res_pjsip_registrar: Remove unavailable contacts if exceeds max_contacts
+ Author: Joseph Nadiv
+ Date: 2021-07-21
+
+ The behavior of max_contacts and remove_existing are connected. If
+ remove_existing is enabled, the soonest expiring contacts are removed.
+ This may occur when there is an unavailable contact. Similarly,
+ when remove_existing is not enabled, registrations from good
+ endpoints are rejected in favor of retaining unavailable contacts.
+
+ This commit adds a new AOR option remove_unavailable, and the effect
+ of this setting will depend on remove_existing. If remove_existing
+ is set to no, we will still remove unavailable contacts when they
+ exceed max_contacts, if there are any. If remove_existing is set to
+ yes, we will prioritize the removal of unavailable contacts before
+ those that are expiring soonest.
+
+ ASTERISK-29525
+
+
+#### ari: Ignore invisible bridges when listing bridges.
+ Author: Joshua C. Colp
+ Date: 2021-09-23
+
+ When listing bridges we go through the ones present in
+ ARI, get their snapshot, turn it into JSON, and add it
+ to the payload we ultimately return.
+
+ An invisible "dial bridge" exists within ARI that would
+ also try to be added to this payload if the channel
+ "create" and "dial" routes were used. This would ultimately
+ fail due to invisible bridges having no snapshot
+ resulting in the listing of bridges failing.
+
+ This change makes it so that the listing of bridges
+ ignores invisible ones.
+
+ ASTERISK-29668
+
+
+#### func_vmcount: Add support for multiple mailboxes
+ Author: Naveen Albert
+ Date: 2021-09-19
+
+ Allows multiple mailboxes to be specified for VMCOUNT
+ instead of just one.
+
+ ASTERISK-29661 #close
+
+
+#### message.c: Support 'To' header override with AMI's MessageSend.
+ Author: Sean Bright
+ Date: 2021-09-21
+
+ The MessageSend AMI action has been updated to allow the Destination
+ and the To addresses to be provided separately. This brings the
+ MessageSend manager command in line with the capabilities of the
+ MessageSend dialplan application.
+
+ ASTERISK-29663 #close
+
+
+#### func_channel: Add CHANNEL_EXISTS function.
+ Author: Naveen Albert
+ Date: 2021-09-15
+
+ Adds a function to check for the existence of a channel by
+ name or by UNIQUEID.
+
+ ASTERISK-29656 #close
+
+
+#### app_queue: Fix hint updates for included contexts
+ Author: Naveen Albert
+ Date: 2021-09-05
+
+ Previously, if custom hints were used with the hint:
+ format in app_queue, when device state changes occured,
+ app_queue would only do a literal string comparison of
+ the context used for the hint in app_queue and the context
+ of the hint which just changed state. This caused hints
+ to not update and become stale if the context associated
+ with the agent included the context which actually changes
+ state, essentially completely breaking device state for
+ any such agents defined in this manner.
+
+ This fix adds an additional check to ensure that included
+ contexts are also compared against the context which changed
+ state, so that the behavior is correct no matter whether the
+ context is specified to app_queue directly or indirectly.
+
+ ASTERISK-29578 #close
+
+
+#### res_http_media_cache.c: Compare unaltered MIME types.
+ Author: Sean Bright
+ Date: 2021-09-10
+
+ Rather than stripping parameters from Content-Type headers before
+ comparison, first try to compare the whole string. If no match is
+ found, strip the parameters and try that way.
+
+ ASTERISK-29275 #close
+
+
+#### logger: Add custom logging capabilities
+ Author: Naveen Albert
+ Date: 2021-07-25
+
+ Adds the ability for users to log to custom log levels
+ by providing custom log level names in logger.conf. Also
+ adds a logger show levels CLI command.
+
+ ASTERISK-29529
+
+
+#### app_externalivr.c: Fix mixed leading whitespace in source code.
+ Author: Sean Bright
+ Date: 2021-09-17
+
+ No functional changes.
+
+
+#### res_rtp_asterisk.c: Fix build failure when not building with pjproject.
+ Author: Guido Falsi
+ Date: 2021-09-17
+
+ Some code has been added referencing symbols defined in a block
+ protected by #ifdef HAVE_PJPROJECT. Protect those code parts in
+ ifdef blocks too.
+
+ ASTERISK-29660
+
+
+#### pjproject: Add patch to fix trailing whitespace issue in rtpmap
+ Author: George Joseph
+ Date: 2021-09-14
+
+ An issue was found where a particular manufacturer's phones add a
+ trailing space to the end of the rtpmap attribute when specifying
+ a payload type that has a "param" after the format name and clock
+ rate. For example:
+
+ a=rtpmap:120 opus/48000/2 \r\n
+
+ Because pjmedia_sdp_attr_get_rtpmap currently takes everything after
+ the second '/' up to the line end as the param, the space is
+ included in future comparisons, which then fail if the param being
+ compared to doesn't also have the space.
+
+ We now use pj_scan_get() to parse the param part of rtpmap so
+ trailing whitespace is automatically stripped.
+
+ ASTERISK-29654
+
+
+#### app_mp3: Force output to 16 bits in mpg123
+ Author: Carlos Oliva
+ Date: 2021-09-13
+
+ In new mpg123 versions (since 1.26) the default output is 32 bits
+ Asterisk expects the output in 16 bits, so we force the output to be on 16 bits.
+ It will work wit new and old versions of mpg123.
+ Thanks Thomas Orgis <thomas-forum@orgis.org> for giving the key!
+
+ ASTERISK-29635 #close
+
+
+#### res_pjsip_caller_id: Add ANI2/OLI parsing
+ Author: Naveen Albert
+ Date: 2021-06-08
+
+ Adds parsing of ANI II digits (Originating
+ Line Information) to PJSIP, on par with
+ what currently exists in chan_sip.
+
+ ASTERISK-29472
+
+
+#### app_mf: Add channel agnostic MF sender
+ Author: Naveen Albert
+ Date: 2021-06-28
+
+ Adds a SendMF application and PlayMF manager
+ event to send arbitrary R1 MF tones on the
+ current or specified channel.
+
+ ASTERISK-29496
+
+
+#### app_stack: Include current location if branch fails
+ Author: Naveen Albert
+ Date: 2021-09-02
+
+ Previously, the error emitted when app_stack tries
+ to branch to a dialplan location that doesn't exist
+ has included only the information about the attempted
+ branch in the error log. This adds the current location
+ as well so users can see where the branch failed in
+ the logs.
+
+ ASTERISK-29626
+
+
+#### test_http_media_cache.c: Fix copy/paste error during test deregistration.
+ Author: Sean Bright
+ Date: 2021-09-10
+
+
+#### resource_channels.c: Fix external media data option
+ Author: Sungtae Kim
+ Date: 2021-09-04
+
+ Fixed the external media creation handle to handle the 'data' option correctly.
+
+ ASTERISK-29629
+
+
+#### func_strings: Add STRBETWEEN function
+ Author: Naveen Albert
+ Date: 2021-09-02
+
+ Adds the STRBETWEEN function, which can be used to insert a
+ substring between each character in a string. For instance,
+ this can be used to insert pauses between DTMF tones in a
+ string of digits.
+
+ ASTERISK-29627
+
+
+#### test_abstract_jb.c: Fix put and put_out_of_order memory leaks.
+ Author: Sean Bright
+ Date: 2021-09-08
+
+ We can't rely on RAII_VAR(...) to properly clean up data that is
+ allocated within a loop.
+
+ ASTERISK-27176 #close
+
+
+#### func_env: Add DIRNAME and BASENAME functions
+ Author: Naveen Albert
+ Date: 2021-09-03
+
+ Adds the DIRNAME and BASENAME functions, which are
+ wrappers around the corresponding C library functions.
+ These can be used to safely and conveniently work with
+ file paths and names in the dialplan.
+
+ ASTERISK-29628 #close
+
+
+#### func_sayfiles: Retrieve say file names
+ Author: Naveen Albert
+ Date: 2021-07-26
+
+ Up until now, all of the logic used to translate
+ arguments to the Say applications has been
+ directly coupled to playback, preventing other
+ modules from using this logic.
+
+ This refactors code in say.c and adds a SAYFILES
+ function that can be used to retrieve the file
+ names that would be played. These can then be
+ used in other applications or for other purposes.
+
+ Additionally, a SayMoney application and a SayOrdinal
+ application are added. Both SayOrdinal and SayNumber
+ are also expanded to support integers greater than
+ one billion.
+
+ ASTERISK-29531
+
+
+#### res_tonedetect: Tone detection module
+ Author: Naveen Albert
+ Date: 2021-08-09
+
+ dsp.c contains arbitrary tone detection functionality
+ which is currently only used for fax tone recognition.
+ This change makes this functionality publicly
+ accessible so that other modules can take advantage
+ of this.
+
+ Additionally, a WaitForTone and TONE_DETECT app and
+ function are included to allow users to do their
+ own tone detection operations in the dialplan.
+
+ ASTERISK-29546
+
+
+#### res_snmp: Add -fPIC to _ASTCFLAGS
+ Author: George Joseph
+ Date: 2021-09-08
+
+ With gcc 11, res/res_snmp.c and res/snmp/agent.c need the
+ -fPIC option added to its _ASTCFLAGS.
+
+ ASTERISK-29634
+
+
+#### app_voicemail.c: Ability to silence instructions if greeting is present.
+ Author: Sean Bright
+ Date: 2021-09-07
+
+ There is an option to silence voicemail instructions but it does not
+ take into consideration if a recorded greeting exists or not. Add a
+ new 'S' option that does that.
+
+ ASTERISK-29632 #close
+
+
+#### term.c: Add support for extended number format terminfo files.
+ Author: Sean Bright
+ Date: 2021-09-04
+
+ ncurses 6.1 introduced an extended number format for terminfo files
+ which the terminfo parsing in Asterisk is not able to parse. This
+ results in some TERM values that do support color (screen-256color on
+ Ubuntu 20.04 for example) to not get a color console.
+
+ ASTERISK-29630 #close
+
+
+#### res_srtp: Disable parsing of not enabled cryptos
+ Author: Jasper Hafkenscheid
+ Date: 2021-09-03
+
+ When compiled without extended srtp crypto suites also disable parsing
+ these from received SDP. This prevents using these, as some client
+ implementations are not stable.
+
+ ASTERISK-29625
+
+
+#### dns.c: Load IPv6 DNS resolvers if configured.
+ Author: Sean Bright
+ Date: 2021-09-06
+
+ IPv6 nameserver addresses are stored in different part of the
+ __res_state structure, so look there if we appear to have support for
+ it.
+
+ ASTERISK-28004 #close
+
+
+#### bridge_softmix: Suppress error on topology change failure
+ Author: George Joseph
+ Date: 2021-09-08
+
+ There are conditions under which a failure to change topology
+ is expected so there's no need to print an ERROR message.
+
+ ASTERISK-29618
+ Reported by: Alexander
+
+
+#### resource_channels.c: Fix wrong external media parameter parse
+ Author: sungtae kim
+ Date: 2021-08-31
+
+ Fixed ARI external media handler to accept body parameters.
+
+ ASTERISK-29622
+
+
+#### config_options: Handle ACO arrays correctly in generated XML docs.
+ Author: Sean Bright
+ Date: 2021-08-25
+
+ There are 3 separate changes here but they are all closely related:
+
+ * Only try to set matchfield attributes on 'field' nodes
+
+ * We need to adjust how we treat the category pointer based on the
+ value of the category_match, to avoid memory corruption. We now
+ generate a regex-like string when match types other than
+ ACO_WHITELIST and ACO_BLACKLIST are used.
+
+ * Switch app_agent_pool from ACO_BLACKLIST_ARRAY to
+ ACO_BLACKLIST_EXACT since we only have one category we need to
+ ignore, not two.
+
+ ASTERISK-29614 #close
+
+
+#### chan_iax2: Add ANI2/OLI information element
+ Author: Naveen Albert
+ Date: 2021-08-18
+
+ Adds an information element for ANI2 so that
+ Originating Line Information can be transmitted
+ over IAX2 channels.
+
+ ASTERISK-29605 #close
+
+
+#### pbx_ael: Fix crash and lockup issue regarding 'ael reload'
+ Author: Mark Murawski
+ Date: 2021-08-31
+
+ Currently pbx_ael does not check if a reload is currently pending
+ before proceeding with a reload. This can cause multiple threads to
+ operate at the same time on what should be mutex protected data. This
+ change adds protection to reloading to ensure only one ael reload is
+ executing at a time.
+
+ ASTERISK-29609 #close
+
+
+#### app_read: Allow reading # as a digit
+ Author: Naveen Albert
+ Date: 2021-08-25
+
+ Allows for the digit # to be read as a digit,
+ just like any other DTMF digit, as opposed to
+ forcing it to be used as an end of input
+ indicator. The default behavior remains
+ unchanged.
+
+ ASTERISK-18454 #close
+
+
+#### res_rtp_asterisk: Automatically refresh stunaddr from DNS
+ Author: Sebastien Duthil
+ Date: 2021-04-05
+
+ This allows the STUN server to change its IP address without having to
+ reload the res_rtp_asterisk module.
+
+ The refresh of the name resolution occurs first when the module is
+ loaded, then recurringly, slightly after the previous DNS answer TTL
+ expires.
+
+ ASTERISK-29508 #close
+
+
+#### bridge_basic: Change warning to verbose if transfer cancelled
+ Author: Naveen Albert
+ Date: 2021-08-25
+
+ The attended transfer feature will emit a warning if the user
+ cancels the transfer or the attended transfer doesn't complete
+ for any reason. Changes the warning to a verbose message,
+ since nothing is actually wrong here.
+
+ ASTERISK-29612 #close
+
+
+#### app_queue: Don't reset queue stats on reload
+ Author: Naveen Albert
+ Date: 2021-08-20
+
+ Prevents reloads of app_queue from also resetting
+ queue statistics.
+
+ Also preserves individual queue agent statistics
+ if we're just reloading members.
+
+ ASTERISK-28701
+
+
+#### res_rtp_asterisk: sqrt(.) requires the header math.h.
+ Author: Alexander Traud
+ Date: 2021-08-25
+
+ ASTERISK-29616
+
+
+#### dialplan: Add one static and fix two whitespace errors.
+ Author: Alexander Traud
+ Date: 2021-08-25
+
+
+#### sig_analog: Changes to improve electromechanical signalling compatibility
+ Author: Sarah Autumn
+ Date: 2021-06-19
+
+ This changeset is intended to address compatibility issues encountered
+ when interfacing Asterisk to electromechanical telephone switches that
+ implement ANI-B, ANI-C, or ANI-D.
+
+ In particular the behaviours that this impacts include:
+
+ - FGC-CAMA did not work at all when using MF signaling. Modified the
+ switch case block to send calls to the correct part of the
+ signaling-handling state machine.
+
+ - For FGC-CAMA operation, the delay between called number ST and
+ second wink for ANI spill has been made configurable; previously
+ all calls were made to wait for one full second.
+
+ - After the ANI spill, previous behavior was to require a 'ST' tone
+ to advance the call. This has been changed to allow 'STP' 'ST2P'
+ or 'ST3P' as well, for compatibility with ANI-D.
+
+ - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable.
+
+ - For calls with an ANI failure, No. 1 Crossbar switches will send
+ forward a single-digit failure code, with no calling number digits
+ and no ST pulse to terminate the spill. I've made the ANI timeout
+ configurable so to reduce dead air time on calls with ANI fail.
+
+ - ANI info digits configurable. Modern digital switches will send 2
+ digits, but ANI-B sends only a single info digit. This caused the
+ ANI reported by Asterisk to be misaligned.
+
+ - Changed a confusing log message to be more informative.
+
+ ASTERISK-29518
+
+
+#### media_cache: Don't lock when curl the remote file
+ Author: Andre Barbosa
+ Date: 2021-08-05
+
+ When playing a remote sound file, which is not in cache, first we need
+ to download it with ast_bucket_file_retrieve.
+
+ This can take a while if the remote host is slow. The current CURL
+ timeout is 180secs, so in extreme situations, it can take 3 minutes to
+ return.
+
+ Because ast_media_cache_retrieve has a lock on all function, while we
+ are waiting for the delayed download, Asterisk is not able to play any
+ more files, even the files already cached locally.
+
+ ASTERISK-29544 #close
+
+
+#### res_pjproject: Allow mapping to Asterisk TRACE level
+ Author: George Joseph
+ Date: 2021-08-16
+
+ Allow mapping pjproject log messages to the Asterisk TRACE
+ log level. The defaults were also changes to log pjproject
+ levels 3,4 to DEBUG and 5,6 to TRACE. Previously 3,4,5,6
+ all went to DEBUG.
+
+ ASTERISK-29582
+
+
+#### app_milliwatt: Timing fix
+ Author: Naveen Albert
+ Date: 2021-08-12
+
+ The Milliwatt application uses incorrect tone timings
+ that cause it to play the 1004 Hz tone constantly.
+
+ This adds an option to enable the correct timing
+ behavior, so that the Milliwatt application can
+ be used for milliwatt test lines. The default behavior
+ remains unchanged for compatability reasons, even
+ though it is incorrect.
+
+ ASTERISK-29575 #close
+
+
+#### func_math: Return integer instead of float if possible
+ Author: Naveen Albert
+ Date: 2021-06-28
+
+ The MIN, MAX, and ABS functions all support float
+ arguments, but currently return floats even if the
+ arguments are all integers and the response is
+ a whole number, in which case the user is likely
+ expecting an integer. This casts the float to an integer
+ before printing into the response buffer if possible.
+
+ ASTERISK-29495
+
+
+#### app_morsecode: Add American Morse code
+ Author: Naveen Albert
+ Date: 2021-08-04
+
+ Previously, the Morsecode application only supported international
+ Morse code. This adds support for American Morse code and adds an
+ option to configure the frequency used in off intervals.
+
+ Additionally, the application checks for hangup between tones
+ to prevent application execution from continuing after hangup.
+
+ ASTERISK-29541
+
+
+#### func_scramble: Audio scrambler function
+ Author: Naveen Albert
+ Date: 2021-08-04
+
+ Adds a function to scramble audio on a channel using
+ whole spectrum frequency inversion. This can be used
+ as a privacy enhancement with applications like
+ ChanSpy or other potentially sensitive audio.
+
+ ASTERISK-29542
+
+
+#### app_originate: Add ability to set codecs
+ Author: Naveen Albert
+ Date: 2021-08-05
+
+ A list of codecs to use for dialplan-originated calls can
+ now be specified in Originate, similar to the ability
+ in call files and the manager action.
+
+ Additionally, we now default to just using the slin codec
+ for originated calls, rather than all the slin* codecs up
+ through slin192, which has been known to cause issues
+ and inconsistencies from AMI and call file behavior.
+
+ ASTERISK-29543
+
+
+#### BuildSystem: Remove two dead exceptions for compiler Clang.
+ Author: Alexander Traud
+ Date: 2021-08-16
+
+ Commit 305ce3d added -Wno-parentheses-equality to Makefile.rules,
+ turning the previous two warning suppressions from commit e9520db
+ redundant. Let us remove the latter.
+
+
+#### chan_alsa, chan_sip: Add replacement to moduleinfo
+ Author: Naveen Albert
+ Date: 2021-08-16
+
+ Adds replacement modules to the moduleinfo for
+ chan_alsa and chan_sip.
+
+ ASTERISK-29601 #close
+
+
+#### res_monitor: Disable building by default.
+ Author: Joshua C. Colp
+ Date: 2021-08-17
+
+ ASTERISK-29602
+
+
+#### muted: Remove deprecated application.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29600
+
+
+#### conf2ael: Remove deprecated application.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29599
+
+
+#### res_config_sqlite: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29598
+
+
+#### chan_vpb: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29597
+
+
+#### chan_misdn: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29596
+
+
+#### chan_nbs: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29595
+
+
+#### chan_phone: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29594
+
+
+#### chan_oss: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29593
+
+
+#### cdr_syslog: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29592
+
+
+#### app_dahdiras: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29591
+
+
+#### app_nbscat: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29590
+
+
+#### app_image: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29589
+
+
+#### app_url: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29588
+
+
+#### app_fax: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29587
+
+
+#### app_ices: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29586
+
+
+#### app_mysql: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29585
+
+
+#### cdr_mysql: Remove deprecated module.
+ Author: Joshua C. Colp
+ Date: 2021-08-16
+
+ ASTERISK-29584
+
+
+#### mgcp: Remove dead debug code
+ Author: Sean Bright
+ Date: 2021-08-10
+
+ ASTERISK-20339 #close
+
+
+#### policy: Deprecate modules and add versions to others.
+ Author: Joshua C. Colp
+ Date: 2021-08-11
+
+ app_meetme is deprecated in 19, to be removed in 21.
+ app_osplookup is deprecated in 19, to be removed in 21.
+ chan_alsa is deprecated in 19, to be removed in 21.
+ chan_mgcp is deprecated in 19, to be removed in 21.
+ chan_skinny is deprecated in 19, to be removed in 21.
+ res_pktccops is deprecated in 19, to be removed in 21.
+ app_macro was deprecated in 16, to be removed in 21.
+ chan_sip was deprecated in 17, to be removed in 21.
+ res_monitor was deprecated in 16, to be removed in 21.
+
+ ASTERISK-29548
+ ASTERISK-29549
+ ASTERISK-29550
+ ASTERISK-29551
+ ASTERISK-29552
+ ASTERISK-29553
+ ASTERISK-29558
+ ASTERISK-29567
+ ASTERISK-29572
+
+
+#### func_frame_drop: New function
+ Author: Naveen Albert
+ Date: 2021-06-16
+
+ Adds function to selectively drop specified frames
+ in the TX or RX direction on a channel, including
+ control frames.
+
+ ASTERISK-29478
+
+
+#### aelparse: Accept an included context with timings.
+ Author: Alexander Traud
+ Date: 2021-08-02
+
+ With Asterisk 1.6.0, in the main parser for the configuration file
+ extensions.conf, the separator was changed from vertical bar to comma.
+ However, the first separator was not changed in aelparse; it still had
+ to be a vertical bar, and no comma was allowed.
+
+ Additionally, this change allows the vertical bar for the first and
+ last parameter again, even in the main parser, because the vertical bar
+ was still accepted for the other parameters.
+
+ ASTERISK-29540
+
+
+#### format_ogg_speex: Implement a "not supported" write handler
+ Author: Kevin Harwell
+ Date: 2021-08-03
+
+ This format did not specify a "write" handler, so when attempting to write
+ to it (ast_writestream) a crash would occur.
+
+ This patch adds a default handler that simply issues a "not supported"
+ warning, thus no longer crashing.
+
+ ASTERISK-29539
+
+
+#### cdr_adaptive_odbc: Prevent filter warnings
+ Author: Naveen Albert
+ Date: 2021-06-28
+
+ Previously, if CDR filters were used so that
+ not all CDR records used all sections defined
+ in cdr_adaptive_odbc.conf, then warnings will
+ always be emitted (if each CDR record is unique
+ to a particular section, n-1 warnings to be
+ specific).
+
+ This turns the offending warning log into
+ a verbose message like the other one, since
+ this behavior is intentional and not
+ indicative of anything wrong.
+
+ ASTERISK-29494
+
+
+#### app_queue: Allow streaming multiple announcement files
+ Author: Naveen Albert
+ Date: 2021-07-25
+
+ Allows multiple files comprising an agent announcement
+ to be played by separating on the ampersand, similar
+ to the multi-file support in other Asterisk applications.
+
+ ASTERISK-29528
+
+
+#### res_pjsip_header_funcs: Add PJSIP_HEADERS() ability to read header by pattern
+ Author: Igor Goncharovsky
+ Date: 2021-04-13
+
+ PJSIP currently does not provide a function to replace SIP_HEADERS() function to get a list of headers from INVITE request.
+ It may be used to get all X- headers in case the actual set and names of headers unknown.
+
+ ASTERISK-29389
+
+
+#### res_statsd: handle non-standard meter type safely
+ Author: Rijnhard Hessel
+ Date: 2021-07-08
+
+ Meter types are not well supported,
+ lacking support in telegraf, datadog and the official statsd servers.
+ We deprecate meters and provide a compliant fallback for any existing usages.
+
+ A flag has been introduced to allow meters to fallback to counters.
+
+
+ ASTERISK-29513
+
+
+#### app_dtmfstore: New application to store digits
+ Author: Naveen Albert
+ Date: 2021-06-16
+
+ Adds application to asynchronously collect digits
+ dialed on a channel in the TX or RX direction
+ using a framehook and stores them in a specified
+ variable, up to a configurable number of digits.
+
+ ASTERISK-29477
+
+
+#### codec_builtin.c: G729 audio gets corrupted by Asterisk due to smoother
+ Author: under
+ Date: 2021-07-22
+
+ If Asterisk gets G.729 6-byte VAD frames inbound, then at outbound Asterisk sends this G.729 stream with non-continuous timestamps.
+ This makes the audio stream not-playable at the receiver side.
+ Linphone isn't able to play such an audio - lots of disruptions are heard.
+ Also I had complains of bad audio from users which use other types of phones.
+
+ After debugging, I found this is a regression connected with RTP Smoother (main/smoother.c).
+
+ Smoother has a special code to handle G.729 VAD frames (search for AST_SMOOTHER_FLAG_G729 in smoother.c).
+
+ However, this flag is never set in Asterisk-12 and newer.
+ Previously it has been set (see Asterisk-11).
+
+ ASTERISK-29526 #close
+
+
+#### res_http_media_cache: Cleanup audio format lookup in HTTP requests
+ Author: Sean Bright
+ Date: 2021-07-23
+
+ Asterisk first looks at the end of the URL to determine the file
+ extension of the returned audio, which in many cases will not work
+ because the URL may end with a query string or a URL fragment. If that
+ fails, Asterisk then looks at the Content-Type header and then finally
+ parses the URL to get the extension.
+
+ The order has been changed such that we look at the Content-Type
+ header first, followed by looking for the extension of the parsed
+ URL. We no longer look at the end of the URL, which was error prone.
+
+ ASTERISK-29527 #close
+
+
+#### docs: Remove embedded macro in WaitForCond XML documentation.
+ Author: Joshua C. Colp
+ Date: 2021-07-27
+
+
+#### Update AMI and ARI versions for Asterisk 20.
+ Author: Ben Ford
+ Date: 2021-07-21
+
+ Bumped AMI and ARI versions for the next major Asterisk version (20).
+
+
+#### AST-2021-009 - pjproject-bundled: Avoid crash during handshake for TLS
+ Author: Kevin Harwell
+ Date: 2021-06-14
+
+ If an SSL socket parent/listener was destroyed during the handshake,
+ depending on timing, it was possible for the handling callback to
+ attempt access of it after the fact thus causing a crash.
+
+ ASTERISK-29415 #close
+
+
+#### AST-2021-008 - chan_iax2: remote crash on unsupported media format
+ Author: Kevin Harwell
+ Date: 2021-05-10
+
+ If chan_iax2 received a packet with an unsupported media format, for
+ example vp9, then it would set the frame's format to NULL. This could
+ then result in a crash later when an attempt was made to access the
+ format.
+
+ This patch makes it so chan_iax2 now ignores/drops frames received
+ with unsupported media format types.
+
+ ASTERISK-29392 #close
+
+
+#### AST-2021-007 - res_pjsip_session: Don't offer if no channel exists.
+ Author: Joshua C. Colp
+ Date: 2021-04-28
+
+ If a re-INVITE is received after we have sent a BYE request then it
+ is possible for no channel to be present on the session. If this
+ occurs we allow PJSIP to produce the offer instead. Since the call
+ is being hung up if it produces an incorrect offer it doesn't
+ actually matter. This also ensures that code which produces SDP
+ does not need to handle if a channel is not present.
+
+ ASTERISK-29381
+
+
+#### res_stasis_playback: Check for chan hangup on play_on_channels
+ Author: Andre Barbosa
+ Date: 2021-06-29
+
+ Verify `ast_check_hangup` before looping to the next sound file.
+ If the call is already hangup we just break the cycle.
+ It also ensures that the PlaybackFinished event is sent if the call was hangup.
+
+ This is also use-full when we are playing a big list of file for a channel that is hangup.
+ Before this patch Asterisk will give a warning for every sound not played and fire a PlaybackStart for every sound file on the list tried to be played.
+
+ With the patch we just break the playback cycle when the chan is hangup.
+
+ ASTERISK-29501 #close
+
+
+#### res_http_media_cache.c: Fix merge errors from 18 -> master
+ Author: Sean Bright
+ Date: 2021-07-02
+
+ ASTERISK-27871 #close
+
+
+#### res_pjsip_stir_shaken: RFC 8225 compliance and error message cleanup.
+ Author: Sean Bright
+ Date: 2021-07-15
+
+ From RFC 8225 Section 5.2.1:
+
+ The "dest" claim is a JSON object with the claim name of "dest"
+ and MUST have at least one identity claim object. The "dest"
+ claim value is an array containing one or more identity claim JSON
+ objects representing the destination identities of any type
+ (currently "tn" or "uri"). If the "dest" claim value array
+ contains both "tn" and "uri" claim names, the JSON object should
+ list the "tn" array first and the "uri" array second. Within the
+ "tn" and "uri" arrays, the identity strings should be put in
+ lexicographical order, including the scheme-specific portion of
+ the URI characters.
+
+ Additionally, make it clear that there was a failure to sign the JWT
+ payload and not necessarily a memory allocation failure.
+
+
+#### res_http_media_cache.c: Parse media URLs to find extensions.
+ Author: Sean Bright
+ Date: 2021-07-02
+
+ Use cURL's URL parsing API, falling back to the urlparser library, to
+ parse playback URLs in order to find their file extensions.
+
+ For backwards compatibility, we first look at the full URL, then at
+ any Content-Type header, and finally at just the path portion of the
+ URL.
+
+ ASTERISK-27871 #close
+
+
+#### main/cdr.c: Correct Party A selection.
+ Author: Sean Bright
+ Date: 2021-07-13
+
+ This appears to just have been a copy/paste error from 6258bbe7. Fix
+ suggested by Ross Beer in ASTERISK~29166.
+
+
+#### stun: Emit warning message when STUN request times out
+ Author: Sebastien Duthil
+ Date: 2021-06-30
+
+ Without this message, it is not obvious that the reason is STUN timeout.
+
+ ASTERISK-29507 #close
+
+
+#### app_reload: New Reload application
+ Author: Naveen Albert
+ Date: 2021-05-26
+
+ Adds an application to reload modules
+ from within the dialplan.
+
+ ASTERISK-29454
+
+
+#### res_ari: Fix audiosocket segfault
+ Author: Igor Goncharovsky
+ Date: 2021-07-08
+
+ Add check that data parameter specified when audiosocket used for externalMedia.
+
+ ASTERISK-29514 #close
+
+
+#### res_pjsip_config_wizard.c: Add port matching support.
+ Author: Sean Bright
+ Date: 2021-06-30
+
+ In f8b0c2c9 we added support for port numbers in 'match' statements
+ but neglected to include that support in the PJSIP config wizard.
+
+ The removed code would have also prevented IPv6 addresses from being
+ successfully used in the config wizard as well.
+
+ ASTERISK-29503 #close
+
+
+#### app_waitforcond: New application
+ Author: Naveen Albert
+ Date: 2021-05-22
+
+ While several applications exist to wait for
+ a certain event to occur, none allow waiting
+ for any generic expression to become true.
+ This application allows for waiting for a condition
+ to become true, with configurable timeout and
+ checking interval.
+
+ ASTERISK-29444
+
+
+#### res_stasis_playback: Send PlaybackFinish event only once for errors
+ Author: Andre Barbosa
+ Date: 2021-06-04
+
+ When we try to play a list of sound files in the same Play command,
+ we get only one PlaybackFinish event, after all sounds are played.
+
+ But in the case where the Play fails (because channel is destroyed
+ for example), Asterisk will send one PlaybackFinish event for each
+ sound file still to be played. If the list is big, Asterisk is
+ sending many events.
+
+ This patch adds a failed state so we can understand that the play
+ failed. On that case we don't send the event, if we still have a
+ list of sounds to be played.
+
+ When we reach the last sound, we send the PlaybackFinish with
+ the failed state.
+
+ ASTERISK-29464 #close
+
+
+#### jitterbuffer: Correct signed/unsigned mismatch causing assert
+ Author: George Joseph
+ Date: 2021-06-17
+
+ If the system time has stepped backwards because of a time
+ adjustment between the time a frame is timestamped and the
+ time we check the timestamps in abstract_jb:hook_event_cb(),
+ we get a negative interval, but we don't check for that there.
+ abstract_jb:hook_event_cb() then calls
+ fixedjitterbuffer:fixed_jb_get() (via abstract_jb:jb_get_fixed)
+ and the first thing that does is assert(interval >= 0).
+
+ There are several issues with this...
+
+ * abstract_jb:hook_event_cb() saves the interval in a variable
+ named "now" which is confusing in itself.
+
+ * "now" is defined as an unsigned int which converts the negative
+ value returned from ast_tvdiff_ms() to a large positive value.
+
+ * fixed_jb_get()'s parameter is defined as a signed int so the
+ interval gets converted back to a negative value.
+
+ * fixed_jb_get()'s assert is NOT an ast_assert but a direct define
+ that points to the system assert() so it triggers even in
+ production mode.
+
+ So...
+
+ * hook_event_cb()'s "now" was renamed to "relative_frame_start" and
+ changed to an int64_t.
+ * hook_event_cb() now checks for a negative value right after
+ retrieving both the current and framedata timestamps and just
+ returns the frame if the difference is negative.
+ * fixed_jb_get()'s local define of ASSERT() was changed to call
+ ast_assert() instead of the system assert().
+
+ ASTERISK-29480
+ Reported by: Dan Cropp
+
+
+#### app_dial: Expanded A option to add caller announcement
+ Author: Naveen Albert
+ Date: 2021-05-21
+
+ Hitherto, the A option has made it possible to play
+ audio upon answer to the called party only. This option
+ is expanded to allow for playback of an audio file to
+ the caller instead of or in addition to the audio
+ played to the answerer.
+
+ ASTERISK-29442
+
+
+#### core: Don't play silence for Busy() and Congestion() applications.
+ Author: Joshua C. Colp
+ Date: 2021-06-21
+
+ When using the Busy() and Congestion() applications the
+ function ast_safe_sleep is used by wait_for_hangup to safely
+ wait on the channel. This function may send silence if Asterisk
+ is configured to do so using the transmit_silence option.
+
+ In a scenario where an answered channel dials a Local channel
+ either directly or through call forwarding and the Busy()
+ or Congestion() dialplan applications were executed with the
+ transmit_silence option enabled the busy or congestion
+ tone would not be heard.
+
+ This is because inband generation of tones (such as busy
+ and congestion) is stopped when other audio is sent to
+ the channel they are being played to. In the given
+ scenario the transmit_silence option would result in
+ silence being sent to the channel, thus stopping the
+ inband generation.
+
+ This change adds a variant of ast_safe_sleep which can be
+ used when silence should not be played to the channel. The
+ wait_for_hangup function has been updated to use this
+ resulting in the tones being generated as expected.
+
+ ASTERISK-29485
+
+
+#### res_pjsip_sdp_rtp: Evaluate remotely held for Session Progress
+ Author: Bernd Zobl
+ Date: 2021-05-07
+
+ With the fix for ASTERISK_28754 channels are no longer put on hold if an
+ outbound INVITE is answered with a "Session Progress" containing
+ "inactive" audio.
+
+ The previous change moved the evaluation of the media attributes to
+ `negotiate_incoming_sdp_stream()` to have the `remotely_held` status
+ available when building the SDP in `create_outgoing_sdp_stream()`.
+ This however means that an answer to an outbound INVITE, which does not
+ traverse `negotiate_incoming_sdp_stream()`, cannot set the
+ `remotely_held` status anymore.
+
+ This change moves the check so that both, `negotiate_incoming_sdp_stream()` and
+ `apply_negotiated_sdp_stream()` can do the checks.
+
+ ASTERISK-29479
+
+
+#### res_pjsip_messaging: Overwrite user in existing contact URI
+ Author: George Joseph
+ Date: 2021-06-16
+
+ When the MessageSend destination is in the form
+ PJSIP/<number>@<endpoint> and the endpoint's contact
+ URI already has a user component, that user component
+ will now be replaced with <number> when creating the
+ request URI.
+
+ ASTERISK_29404
+
+
+#### res_pjsip/pjsip_message_filter: set preferred transport in pjsip_message_filter
+ Author: Bernd Zobl
+ Date: 2021-03-16
+
+ Set preferred transport when querying the local address to use in
+ filter_on_tx_messages(). This prevents the module to erroneously select
+ the wrong transport if more than one transports of the same type (TCP or
+ TLS) are configured.
+
+ ASTERISK-29241
+
+
+#### pbx_builtins: Corrects SayNumber warning
+ Author: Naveen Albert
+ Date: 2021-06-10
+
+ Previously, SayNumber always emitted a warning if the caller hung up
+ during execution. Usually this isn't correct, so check if the channel
+ hung up and, if so, don't emit a warning.
+
+ ASTERISK-29475
+
+
+#### func_lock: Add "dialplan locks show" cli command.
+ Author: Jaco Kroon
+ Date: 2021-05-22
+
+ For example:
+
+ arthur*CLI> dialplan locks show
+ func_lock locks:
+ Name Requesters Owner
+ uls-autoref 0 (unlocked)
+ 1 total locks listed.
+
+ Obviously other potentially useful stats could be added (eg, how many
+ times there was contention, how many times it failed etc ... but that
+ would require keeping the stats and I'm not convinced that's worth the
+ effort. This was useful to troubleshoot some other issues so submitting
+ it.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### func_lock: Prevent module unloading in-use module.
+ Author: Jaco Kroon
+ Date: 2021-05-22
+
+ The scenario where a channel still has an associated datastore we
+ cannot unload since there is a function pointer to the destroy and fixup
+ functions in play. Thus increase the module ref count whenever we
+ allocate a datastore, and decrease it during destroy.
+
+ In order to tighten the race that still exists in spite of this (below)
+ add some extra failure cases to prevent allocations in these cases.
+
+ Race:
+
+ If module ref is zero, an LOCK or TRYLOCK is invoked (near)
+ simultaneously on a channel that has NOT PREVIOUSLY taken a lock, and if
+ in such a case the datastore is created *prior* to unloading being set
+ to true (first step in module unload) then it's possible that the module
+ will unload with the destructor being called (and segfault) post the
+ module being unloaded. The module will however wait for such locks to
+ release prior to unloading.
+
+ If post that we can recheck the module ref before returning the we can
+ (in theory, I think) eliminate the last of the race. This race is
+ mostly theoretical in nature.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### func_lock: Fix memory corruption during unload.
+ Author: Jaco Kroon
+ Date: 2021-05-22
+
+ AST_TRAVERSE accessess current as current = current->(field).next ...
+ and since we free current (and ast_free poisons the memory) we either
+ end up on a ast_mutex_lock to a non-existing lock that can never be
+ obtained, or a segfault.
+
+ Incidentally add logging in the "we have to wait for a lock to release"
+ case, and remove an ineffective statement that sets memory that was just
+ cleared by ast_calloc to zero.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### func_lock: Fix requesters counter in error paths.
+ Author: Jaco Kroon
+ Date: 2021-05-22
+
+ In two places we bail out with failure after we've already incremented
+ the requesters counter, if this occured then it would effectively result
+ in unload to wait indefinitely, thus preventing clean shutdown.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### app_originate: Allow setting Caller ID and variables
+ Author: Naveen Albert
+ Date: 2021-05-25
+
+ Caller ID can now be set on the called channel and
+ Variables can now be set on the destination
+ using the Originate application, just as
+ they can be currently using call files
+ or the Manager Action.
+
+ ASTERISK-29450
+
+
+#### menuselect: Fix description of several modules.
+ Author: Sean Bright
+ Date: 2021-06-10
+
+ The text description needs to be the last thing on the AST_MODULE_INFO
+ line to be pulled in properly by menuselect.
+
+
+#### app_confbridge: New ConfKick() application
+ Author: Naveen Albert
+ Date: 2021-05-23
+
+ Adds a new ConfKick() application, which may
+ be used to kick a specific channel, all channels,
+ or all non-admin channels from a specified
+ conference bridge, similar to existing CLI and
+ AMI commands.
+
+ ASTERISK-29446
+
+
+#### res_pjsip_dtmf_info: Hook flash
+ Author: Naveen Albert
+ Date: 2021-06-02
+
+ Adds hook flash recognition support
+ for application/hook-flash.
+
+ ASTERISK-29460
+
+
+#### app_confbridge: New option to prevent answer supervision
+ Author: Naveen Albert
+ Date: 2021-05-20
+
+ A new user option, answer_channel, adds the capability to
+ prevent answering the channel if it hasn't already been
+ answered yet.
+
+ ASTERISK-29440
+
+
+#### sip_to_pjsip: Fix missing cases
+ Author: Naveen Albert
+ Date: 2021-06-02
+
+ Adds the "auto" case which is valid with
+ both chan_sip dtmfmode and chan_pjsip's
+ dtmf_mode, adds subscribecontext to
+ subscribe_context conversion, and accounts
+ for cipher = ALL being invalid.
+
+ ASTERISK-29459
+
+
+#### res_pjsip_messaging: Refactor outgoing URI processing
+ Author: George Joseph
+ Date: 2021-04-22
+
+ * Implemented the new "to" parameter of the MessageSend()
+ dialplan application. This allows a user to specify
+ a complete SIP "To" header separate from the Request URI.
+
+ * Completely refactored the get_outbound_endpoint() function
+ to actually handle all the destination combinations that
+ we advertized as supporting.
+
+ * We now also accept a destination in the same format
+ as Dial()... PJSIP/number@endpoint
+
+ * Added lots of debugging.
+
+ ASTERISK-29404
+ Reported by Brian J. Murrell
+
+
+#### func_math: Three new dialplan functions
+ Author: Naveen Albert
+ Date: 2021-05-16
+
+ Introduces three new dialplan functions, MIN and MAX,
+ which can be used to calculate the minimum or
+ maximum of up to two numbers, and ABS, an absolute
+ value function.
+
+ ASTERISK-29431
+
+
+#### STIR/SHAKEN: Add Date header, dest->tn, and URL checking.
+ Author: Ben Ford
+ Date: 2021-05-19
+
+ STIR/SHAKEN requires a Date header alongside the Identity header, so
+ that has been added. Still on the outgoing side, we were missing the
+ dest->tn section of the JSON payload, so that has been added as well.
+ Moving to the incoming side, URL checking has been added to the public
+ cert URL to ensure that it starts with http.
+
+ https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
+
+
+#### res_pjsip: On partial transport reload also move factories.
+ Author: Joshua C. Colp
+ Date: 2021-05-24
+
+ For connection oriented transports PJSIP uses factories to
+ produce transports. When doing a partial transport reload
+ we need to also move the factory of the transport over so
+ that anything referencing the transport (such as an endpoint)
+ has the factory available.
+
+ ASTERISK-29441
+
+
+#### func_volume: Add read capability to function.
+ Author: Naveen Albert
+ Date: 2021-05-20
+
+ Up until now, the VOLUME function has been write
+ only, so that TX/RX values can be set but not
+ read afterwards. Now, previously set TX/RX values
+ can be read later.
+
+ ASTERISK-29439
+
+
+#### stasis: Fix "FRACK!, Failed assertion bad magic number" when unsubscribing
+ Author: Evgenios_Greek
+ Date: 2021-04-13
+
+ When unsubscribing from an endpoint technology a FRACK
+ would occur due to incorrect reference counting. This fixes
+ that issue, along with some other issues.
+
+ Fixed a typo in get_subscription when calling ao2_find as it
+ needed to pass the endpoint ID and not the entire object.
+
+ Fixed scenario where a subscription would get returned when
+ it shouldn't have been when searching based on endpoint
+ technology.
+
+ A doulbe unreference has also been resolved by only explicitly
+ releasing the reference held by tech_subscriptions.
+
+ ASTERISK-28237 #close
+ Reported by: Lucas Tardioli Silveira
+
+
+#### res_pjsip.c: Support endpoints with domain info in username
+ Author: Joseph Nadiv
+ Date: 2021-05-20
+
+ In multidomain environments, it is desirable to create
+ PJSIP endpoints with the domain info in the endpoint name
+ in pjsip_endpoint.conf. This resulted in an error with
+ registrations, NOTIFY, and OPTIONS packet generation.
+
+ This commit will detect if there is an @ in the endpoint
+ identifier and generate the URI accordingly so NOTIFY and
+ OPTIONS From headers will generate correctly.
+
+ ASTERISK-28393
+
+
+#### res_rtp_asterisk: Set correct raddr port on RTCP srflx candidates.
+ Author: Joshua C. Colp
+ Date: 2021-05-20
+
+ RTCP ICE candidates use a base address derived from the RTP
+ candidate. The port on the base address was not being updated to
+ the RTCP port.
+
+ This change sets the base port to the RTCP port and all is well.
+
+ ASTERISK-29433
+
+
+#### asterisk: We've moved to Libera Chat!
+ Author: Joshua C. Colp
+ Date: 2021-05-25
+
+
+#### res_rtp_asterisk: make it possible to remove SOFTWARE attribute
+ Author: Jeremy Lainé
+ Date: 2021-05-19
+
+ By default Asterisk reports the PJSIP version in a SOFTWARE attribute
+ of every STUN packet it sends. This may not be desired in a production
+ environment, and RFC5389 recommends making the use of the SOFTWARE
+ attribute a configurable option:
+
+ https://datatracker.ietf.org/doc/html/rfc5389#section-16.1.2
+
+ This patch adds a `stun_software_attribute` yes/no option to make it
+ possible to omit the SOFTWARE attribute from STUN packets.
+
+ ASTERISK-29434
+
+
+#### res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs
+ Author: George Joseph
+ Date: 2021-04-15
+
+ RFC7616 and RFC8760 allow more than one WWW-Authenticate or
+ Proxy-Authenticate header per realm, each with different digest
+ algorithms (including new ones like SHA-256 and SHA-512-256).
+ Thankfully however a UAS can NOT send back multiple Authenticate
+ headers for the same realm with the same digest algorithm. The
+ UAS is also supposed to send the headers in order of preference
+ with the first one being the most preferred. We're supposed to
+ send an Authorization header for the first one we encounter for a
+ realm that we can support.
+
+ The UAS can also send multiple realms, especially when it's a
+ proxy that has forked the request in which case the proxy will
+ aggregate all of the Authenticate headers and then send them all
+ back to the UAC.
+
+ It doesn't stop there though... Each realm can require a
+ different username from the others. There's also nothing
+ preventing each digest algorithm from having a unique password
+ although I'm not sure if that adds any benefit.
+
+ So now... For each Authenticate header we encounter, we have to
+ determine if we support the digest algorithm and, if not, just
+ skip the header. We then have to find an auth object that
+ matches the realm AND the digest algorithm or find a wildcard
+ object that matches the digest algorithm. If we find one, we add
+ it to the results vector and read the next Authenticate header.
+ If the next header is for the same realm AND we already added an
+ auth object for that realm, we skip the header. Otherwise we
+ repeat the process for the next header.
+
+ In the end, we'll have accumulated a list of credentials we can
+ pass to pjproject that it can use to add Authentication headers
+ to a request.
+
+ NOTE: Neither we nor pjproject can currently handle digest
+ algorithms other than MD5. We don't even have a place for it in
+ the ast_sip_auth object. For this reason, we just skip processing
+ any Authenticate header that's not MD5. When we support the
+ others, we'll move the check into the loop that searches the
+ objects.
+
+ Changes:
+
+ * Added a new API ast_sip_retrieve_auths_vector() that takes in
+ a vector of auth ids (usually supplied on a call to
+ ast_sip_create_request_with_auth()) and populates another
+ vector with the actual objects.
+
+ * Refactored res_pjsip_outbound_authenticator_digest to handle
+ multiple Authenticate headers and set the stage for handling
+ additional digest algorithms.
+
+ * Added a pjproject patch that allows them to ignore digest
+ algorithms they don't support. This patch has already been
+ merged upstream.
+
+ * Updated documentation for auth objects in the XML and
+ in pjsip.conf.sample.
+
+ * Although res_pjsip_authenticator_digest isn't affected
+ by this change, some debugging and a testsuite AMI event
+ was added to facilitate testing.
+
+ Discovered during OpenSIPit 2021.
+
+ ASTERISK-29397
+
+
+#### res_pjsip_dialog_info_body_generator: Add LOCAL/REMOTE tags in dialog-info+xml
+ Author: Joseph Nadiv
+ Date: 2021-04-14
+
+ RFC 4235 Section 4.1.6 describes XML elements that should be
+ sent to subscribed endpoints to identify the local and remote
+ participants in the dialog.
+
+ This patch adds this functionality to PJSIP by iterating through the
+ ringing channels causing the NOTIFY, and inserts the channel info
+ into the dialog so that information is properly passed to the endpoint
+ in dialog-info+xml.
+
+ ASTERISK-24601
+ Patch submitted: Joshua Elson
+ Modified by: Joseph Nadiv and Sean Bright
+ Tested by: Joseph Nadiv
+
+
+#### AMI: Add AMI event to expose hook flash events
+ Author: Naveen Albert
+ Date: 2021-05-13
+
+ Although Asterisk can receive and propogate flash events, it currently
+ provides no mechanism for doing anything with them itself.
+
+ This AMI event allows flash events to be processed by Asterisk.
+ Additionally, AST_CONTROL_FLASH is included in a switch statement
+ in channel.c to avoid throwing a warning when we shouldn't.
+
+ ASTERISK-29380
+
+
+#### app_voicemail: Configurable voicemail beep
+ Author: Naveen Albert
+ Date: 2021-05-13
+
+ Hitherto, VoiceMail() played a non-customizable beep tone to indicate
+ the caller could leave a message. In some cases, the beep may not
+ be desired, or a different tone may be desired.
+
+ To increase flexibility, a new option allows customization of the tone.
+ If the t option is specified, the default beep will be overridden.
+ Supplying an argument will cause it to use the specified file for the tone,
+ and omitting it will cause it to skip the beep altogether. If the option
+ is not used, the default behavior persists.
+
+ ASTERISK-29349
+
+
+#### main/file.c: Don't throw error on flash event.
+ Author: Naveen Albert
+ Date: 2021-05-13
+
+ AST_CONTROL_FLASH isn't accounted for in a switch statement in file.c
+ where it should be ignored. Adding this to the switch ensures a
+ warning isn't thrown on RFC2833 flash events, since nothing's amiss.
+
+ ASTERISK-29372
+
+
+#### chan_sip: Expand hook flash recognition.
+ Author: Naveen Albert
+ Date: 2021-05-13
+
+ Some ATAs send hook flash events as application/hook-flash, rather than a DTMF
+ event. Now, we also recognize hook-flash as a flash event.
+
+ ASTERISK-29370
+
+
+#### pjsip: Add patch for resolving STUN packet lifetime issues.
+ Author: Joshua C. Colp
+ Date: 2021-05-11
+
+ In some cases it was possible for a STUN packet to be destroyed
+ prematurely or even destroyed partially multiple times.
+
+ This patch provided by Teluu fixes the lifetime of these
+ packets and ensures they aren't partially destroyed multiple
+ times.
+
+ https://github.com/pjsip/pjproject/pull/2709
+
+ ASTERISK-29377
+
+
+#### chan_pjsip: Correct misleading trace message
+ Author: Sean Bright
+ Date: 2021-05-12
+
+ ASTERISK-29358 #close
+
+
+#### STIR/SHAKEN: Switch to base64 URL encoding.
+ Author: Ben Ford
+ Date: 2021-04-26
+
+ STIR/SHAKEN encodes using base64 URL format. Currently, we just use
+ base64. New functions have been added that convert to and from base64
+ encoding.
+
+ The origid field should also be an UUID. This means there's no reason to
+ have it as an option in stir_shaken.conf, as we can simply generate one
+ when creating the Identity header.
+
+ https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
+
+
+#### STIR/SHAKEN: OPENSSL_free serial hex from openssl.
+ Author: Ben Ford
+ Date: 2021-05-11
+
+ We're getting the serial number of the certificate from openssl and
+ freeing it with ast_free(), but it needs to be freed with OPENSSL_free()
+ instead. Now we duplicate the string and free the one from openssl with
+ OPENSSL_free(), which means we can still use ast_free() on the returned
+ string.
+
+ https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
+
+
+#### STIR/SHAKEN: Fix certificate type and storage.
+ Author: Ben Ford
+ Date: 2021-04-21
+
+ During OpenSIPit, we found out that the public certificates must be of
+ type X.509. When reading in public keys, we use the corresponding X.509
+ functions now.
+
+ We also discovered that we needed a better naming scheme for the
+ certificates since certificates with the same name would cause issues
+ (overwriting certs, etc.). Now when we download a public certificate, we
+ get the serial number from it and use that as the name of the cached
+ certificate.
+
+ The configuration option public_key_url in stir_shaken.conf has also
+ been renamed to public_cert_url, which better describes what the option
+ is for.
+
+ https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021
+
+
+#### translate.c: Avoid refleak when checking for a translation path
+ Author: Sean Bright
+ Date: 2021-04-30
+
+
+#### res_rtp_asterisk: More robust timestamp checking
+ Author: Sean Bright
+ Date: 2021-04-27
+
+ We assume that a timestamp value of 0 represents an 'uninitialized'
+ timestamp, but 0 is a valid value. Add a simple wrapper to be able to
+ differentiate between whether the value is set or not.
+
+ This also removes the fix for ASTERISK~28812 which should not be
+ needed if we are checking the last timestamp appropriately.
+
+ ASTERISK-29030 #close
+
+
+#### chan_local: Skip filtering audio formats on removed streams.
+ Author: Joshua C. Colp
+ Date: 2021-04-28
+
+ When a stream topology is provided to chan_local when dialing
+ it filters the audio formats down. This operation did not skip
+ streams which were removed (that have no formats) resulting in
+ calling being aborted.
+
+ This change causes such streams to be skipped.
+
+ ASTERISK-29407
+
+
+#### res_pjsip.c: OPTIONS processing can now optionally skip authentication
+ Author: Sean Bright
+ Date: 2021-04-23
+
+ ASTERISK-27477 #close
+
+
+#### translate.c: Take sampling rate into account when checking codec's buffer size
+ Author: Jean Aunis
+ Date: 2021-04-21
+
+ Up/down sampling changes the number of samples produced by a translation.
+ This must be taken into account when checking the codec's buffer size.
+
+ ASTERISK-29328
+
+
+#### svn: Switch to https scheme.
+ Author: Joshua C. Colp
+ Date: 2021-04-25
+
+ Some versions of SVN seemingly don't follow the redirect
+ to https.
+
+
+#### res_pjsip: Update documentation for the auth object
+ Author: George Joseph
+ Date: 2021-04-20
+
+
+#### res_aeap: Add basic config skeleton and CLI commands.
+ Author: Ben Ford
+ Date: 2021-03-29
+
+ Added support for a basic AEAP configuration read from aeap.conf.
+ Also added 2 CLI commands for showing individual configurations as
+ well as all of them: aeap show server <id> and aeap show servers.
+
+ Only one configuration option is required at the moment, and that one is
+ server_url. It must be a websocket URL. The other option, codecs, is
+ optional and will be used over the codecs specified on the endpoint if
+ provided.
+
+ https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453
+
+
+#### bridge_channel_write_frame: Check for NULL channel
+ Author: George Joseph
+ Date: 2021-04-02
+
+ There is a possibility, when bridge_channel_write_frame() is
+ called, that the bridge_channel->chan will be NULL. The first
+ thing bridge_channel_write_frame() does though is call
+ ast_channel_is_multistream() which had no check for a NULL
+ channel and therefore caused a segfault. Since it's still
+ possible for bridge_channel_write_frame() to write the frame to
+ the other channels in the bridge, we don't want to bail before we
+ call ast_channel_is_multistream() but we can just skip the
+ multi-channel stuff. So...
+
+ bridge_channel_write_frame() only calls ast_channel_is_multistream()
+ if bridge_channel->chan is not NULL.
+
+ As a safety measure, ast_channel_is_multistream() now returns
+ false if the supplied channel is NULL.
+
+ ASTERISK-29379
+ Reported-by: Vyrva Igor
+ Reported-by: Ross Beer
+
+
+#### loader.c: Speed up deprecation metadata lookup
+ Author: Sean Bright
+ Date: 2021-04-01
+
+ Only use an XPath query once per module, then just navigate the DOM for
+ everything else.
+
+
+#### res_prometheus: Clone containers before iterating
+ Author: George Joseph
+ Date: 2021-04-01
+
+ The channels, bridges and endpoints scrape functions were
+ grabbing their respective global containers, getting the
+ count of entries, allocating metric arrays based on
+ that count, then iterating over the container. If the
+ global container had new objects added after the count
+ was taken and the metric arrays were allocated, we'd run
+ out of metric entries and attempt to write past the end
+ of the arrays.
+
+ Now each of the scape functions clone their respective
+ global containers and all operations are done on the
+ clone. Since the clone is stable between getting the
+ count and iterating over it, we can't run past the end
+ of the metrics array.
+
+ ASTERISK-29130
+ Reported-By: Francisco Correia
+ Reported-By: BJ Weschke
+ Reported-By: Sébastien Duthil
+
+
+#### loader: Output warnings for deprecated modules.
+ Author: Joshua C. Colp
+ Date: 2021-03-10
+
+ Using the information from the MODULEINFO XML we can
+ now output useful information at the end of module
+ loading for deprecated modules. This includes the
+ version it was deprecated in, the version it will be
+ removed in, and the replacement if available.
+
+ ASTERISK-29339
+
+
+#### res_rtp_asterisk: Fix standard deviation calculation
+ Author: Kevin Harwell
+ Date: 2021-03-22
+
+ For some input to the standard deviation algorithm extremely large,
+ and wrong numbers were being calculated.
+
+ This patch uses a new formula for correctly calculating both the
+ running mean and standard deviation for the given inputs.
+
+ ASTERISK-29364 #close
+
+
+#### res_rtp_asterisk: Don't count 0 as a minimum lost packets
+ Author: Kevin Harwell
+ Date: 2021-03-29
+
+ The calculated minimum lost packets represents the lowest number of
+ lost packets missed during an RTCP report interval. Zero of course
+ is the lowest, but the idea is that this value contain the lowest
+ number of lost packets once some have been missed.
+
+ This patch checks to make sure the number of lost packets over an
+ interval is not zero before checking and setting the minimum value.
+
+ Also, this patch updates the rtp lost packet test to check for
+ packet loss over several reports vs one.
+
+
+#### res_rtp_asterisk: Statically declare rtp_drop_packets_data object
+ Author: Kevin Harwell
+ Date: 2021-03-31
+
+ This patch makes the drop_packets_data object static.
+
+
+#### res_rtp_asterisk: Only raise flash control frame on end.
+ Author: Joshua C. Colp
+ Date: 2021-03-29
+
+ Flash in RTP is conveyed the same as DTMF, just with a
+ specific digit. In Asterisk however we do flash as a
+ single control frame.
+
+ This change makes it so that only on end do we provide
+ the flash control frame to the core. Previously we would
+ provide a flash control frame on both begin and end,
+ causing flash to work improperly.
+
+ ASTERISK-29373
+
+
+#### res_rtp_asterisk: Add a DEVMODE RTP drop packets CLI command
+ Author: Kevin Harwell
+ Date: 2021-03-05
+
+ This patch makes it so when Asterisk is compiled in DEVMODE a CLI
+ command is available that allows someone to drop incoming RTP
+ packets. The command allows for dropping of packets once, or on a
+ timed interval (e.g. drop 10 packets every 5 seconds). A user can
+ also specify to drop packets by IP address.
+
+
+#### res_pjsip: Give error when TLS transport configured but not supported.
+ Author: Joshua C. Colp
+ Date: 2021-03-30
+
+
+#### time: Add timeval create and unit conversion functions
+ Author: Kevin Harwell
+ Date: 2021-03-05
+
+ Added a TIME_UNIT enumeration, and a function that converts a
+ string to one of the enumerated values. Also, added functions
+ that create and initialize a timeval object using a specified
+ value, and unit type.
+
+
+#### app_queue: Add alembic migration to add ringinuse to queue_members.
+ Author: Sean Bright
+ Date: 2021-03-24
+
+ ASTERISK-28356 #close
+
+
+#### modules.conf: Fix more differing usages of assignment operators.
+ Author: Sean Bright
+ Date: 2021-03-28
+
+ I missed the changes in 18 and master in the previous review.
+
+ ASTERISK-24434 #close
+
+
+#### logger.conf.sample: Add more debug documentation.
+ Author: Ben Ford
+ Date: 2021-03-24
+
+
+#### logging: Add .log to samples and update asterisk.logrotate.
+ Author: Ben Ford
+ Date: 2021-03-24
+
+ Added .log extension to the sample logs in logger.conf.sample so that
+ they will be able to be opened in the browser when attached to JIRA
+ tickets. Because of this, asterisk.logrotate has also been updated to
+ look for .log extensions instead of no extension for log files such as
+ full and messages.
+
+
+#### app_queue.c: Remove dead 'updatecdr' code.
+ Author: Sean Bright
+ Date: 2021-03-23
+
+ Also removed the sample documentation, and some oddly-placed
+ documentation about the timeout argument to the Queue() application
+ itself. There is a large section on the timeout behavior below.
+
+ ASTERISK-26614 #close
+
+
+#### queues.conf.sample: Correct 'context' documentation.
+ Author: Sean Bright
+ Date: 2021-03-23
+
+ ASTERISK-24631 #close
+
+
+#### logger: Console sessions will now respect logger.conf dateformat= option
+ Author: Mark Murawski
+ Date: 2021-03-19
+
+ The 'core' console (ie: asterisk -c) does read logger.conf and does
+ use the dateformat= option.
+
+ Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf
+ and uses a hard coded dateformat option for printing received verbose messages:
+ main/logger.c: static char dateformat[256] = "%b %e %T"
+
+ This change will load logger.conf for each remote console session and
+ use the dateformat= option to set the per-line timestamp for verbose messages
+
+ ASTERISK-25358: #close
+ Reported-by: Igor Liferenko
+
+#### app_queue.c: Don't crash when realtime queue name is empty.
+ Author: Sean Bright
+ Date: 2021-03-19
+
+ ASTERISK-27542 #close
+
+
+#### res_pjsip_session: Make reschedule_reinvite check for NULL topologies
+ Author: George Joseph
+ Date: 2021-03-18
+
+ When the check for equal topologies was added to reschedule_reinvite()
+ it was assumed that both the pending and active media states would
+ actually have non-NULL topologies. We since discovered this isn't
+ the case.
+
+ We now only test for equal topologies if both media states have
+ non-NULL topologies. The logic had to be rearranged a bit to make
+ sure that we cloned the media states if their topologies were
+ non-NULL but weren't equal.
+
+ ASTERISK-29215
+
+
+#### app_queue: Only send QueueMemberStatus if status changes.
+ Author: Joshua C. Colp
+ Date: 2021-03-19
+
+ If a queue member was updated with the same status multiple
+ times each time a QueueMemberStatus event would be sent
+ which would be a duplicate of the previous.
+
+ This change makes it so that the QueueMemberStatus event is
+ only sent if the status actually changes.
+
+ ASTERISK-29355
+
+
+#### core_unreal: Fix deadlock with T.38 control frames.
+ Author: Joshua C. Colp
+ Date: 2021-03-19
+
+ When using the ast_unreal_lock_all function no channel
+ locks can be held before calling it.
+
+ This change unlocks the channel that indicate was
+ called on before doing so and then relocks it afterwards.
+
+ ASTERISK-29035
+
+
+#### res_pjsip: Add support for partial transport reload.
+ Author: Joshua C. Colp
+ Date: 2021-03-01
+
+ Some configuration items for a transport do not result in
+ the underlying transport changing, but instead are just
+ state we keep ourselves and use. It is perfectly reasonable
+ to change these items.
+
+ These include local_net and external_* information.
+
+ ASTERISK-29354
+
+
+#### menuselect: exit non-zero in case of failure on --enable|disable options.
+ Author: Jaco Kroon
+ Date: 2021-03-13
+
+ ASTERISK-29348
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### res_rtp_asterisk: Force resync on SSRC change.
+ Author: Joshua C. Colp
+ Date: 2021-03-17
+
+ When an SSRC change occurs the timestamps are likely
+ to change as well. As a result we need to reset the
+ timestamp mapping done in the calc_rxstamp function
+ so that they map properly from timestamp to real
+ time.
+
+ This previously occurred but due to packet
+ retransmission support the explicit setting
+ of the marker bit was not effective.
+
+ ASTERISK-29352
+
+
+#### menuselect: Add ability to set deprecated and removed versions.
+ Author: Joshua C. Colp
+ Date: 2021-03-10
+
+ The "deprecated_in" and "removed_in" information can now be
+ set in MODULEINFO for a module and is then displayed in
+ menuselect so users can be aware of when a module is slated
+ to be deprecated and then removed.
+
+ ASTERISK-29337
+
+
+#### xml: Allow deprecated_in and removed_in for MODULEINFO.
+ Author: Joshua C. Colp
+ Date: 2021-03-10
+
+ ASTERISK-29337
+
+
+#### xml: Embed module information into core XML documentation.
+ Author: Joshua C. Colp
+ Date: 2021-03-09
+
+ This change embeds the MODULEINFO block of modules
+ into the core XML documentation. This provides a shared
+ mechanism for use by both menuselect and Asterisk for
+ information and a definitive source of truth.
+
+ ASTERISK-29335
+
+
+#### documentation: Fix non-matching module support levels.
+ Author: Joshua C. Colp
+ Date: 2021-03-10
+
+ Some modules have a different support level documented in their
+ MODULEINFO XML and Asterisk module definition. This change
+ brings the two in sync for the modules which were not matching.
+
+ ASTERISK-29336
+
+
+#### channel: Fix crash in suppress API.
+ Author: Joshua C. Colp
+ Date: 2021-03-09
+
+ There exists an inconsistency with framehook usage
+ such that it is only on reads that the frame should
+ be freed, not on writes as well.
+
+ ASTERISK-29071
+
+
+#### func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds
+ Author: Jaco Kroon
+ Date: 2021-02-24
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### app.h: Fix -Werror=zero-length-bounds compile errors in dev mode.
+ Author: Jaco Kroon
+ Date: 2021-02-24
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### app_dial.c: Only send DTMF on first progress event.
+ Author: Sean Bright
+ Date: 2021-03-06
+
+ ASTERISK-29329 #close
+
+
+#### res_format_attr_*: Parameter Names are Case-Insensitive.
+ Author: Alexander Traud
+ Date: 2021-03-05
+
+ see RFC 4855:
+ parameter names are case-insensitive both in media type strings and
+ in the default mapping to the SDP a=fmtp attribute.
+
+ This change is required for H.263+ because some implementations are
+ known to use even mixed-case. This does not fix ASTERISK~29268 because
+ H.264 was not fixed. This approach here lowers/uppers both parameter
+ names and parameter values. H.264 needs a different approach because
+ one of its parameter values is not case-insensitive:
+ sprop-parameter-sets is Base64.
+
+
+#### chan_iax2: System Header strings is included via asterisk.h/compat.h.
+ Author: Alexander Traud
+ Date: 2021-03-05
+
+ The system header strings was included mistakenly with commit 3de0204.
+ That header is included via asterisk.h and there via the compat.h.
+
+
+#### modules.conf: Fix differing usage of assignment operators.
+ Author: Sean Bright
+ Date: 2021-03-08
+
+ ASTERISK-24434 #close
+
+
+#### strings.h: ast_str_to_upper() and _to_lower() are not pure.
+ Author: Sean Bright
+ Date: 2021-03-08
+
+ Because they modify their argument they are not pure functions and
+ should not be marked as such, otherwise the compiler may optimize
+ them away.
+
+ ASTERISK-29306 #close
+
+
+#### res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse.
+ Author: Sean Bright
+ Date: 2021-03-08
+
+ ao2_replace() bumps the reference count of the object that is doing the
+ replacing, which is not what we want. We just want to drop the old ref
+ on the old object and update the pointer to point to the new object.
+
+ Pointed out by George Joseph in #asterisk-dev
+
+
+#### res/res_rtp_asterisk: generate new SSRC on native bridge end
+ Author: Torrey Searle
+ Date: 2021-02-19
+
+ For RTCP to work, we update the ssrc to be the one corresponding to
+ the native bridge while active. However when the bridge ends we
+ should generate a new SSRC as the sequence numbers will not continue
+ from the native bridge left off.
+
+ ASTERISK-29300 #close
+
+
+#### sorcery: Add support for more intelligent reloading.
+ Author: Joshua C. Colp
+ Date: 2021-03-01
+
+ Some sorcery objects actually contain dynamic content
+ that can change despite the underlying configuration
+ itself not changing. A good example of this is the
+ res_pjsip_endpoint_identifier_ip module which allows
+ specifying hostnames. While the configuration may not
+ change between reloads the DNS information of the
+ hostnames can.
+
+ This change adds the ability for a sorcery object to be
+ marked as having dynamic contents which is then taken
+ into account when reloading by the sorcery file based
+ config module. If there is an object with dynamic content
+ then a reload will be forced while if there are none
+ then the existing behavior of not reloading occurs.
+
+ ASTERISK-29321
+
+
+#### res_pjsip_refer: Move the progress dlg release to a serializer
+ Author: George Joseph
+ Date: 2021-03-02
+
+ Although the dlg session count was incremented in a pjsip servant
+ thread, there's no guarantee that the last thread to unref this
+ progress object was one. Before we decrement, we need to make
+ sure that this is either a servant thread or that we push the
+ decrement to a serializer that is one.
+
+ Because pjsip_dlg_dec_session requires the dialog lock, we don't
+ want to wait on the task to complete if we had to push it to a
+ serializer.
+
+
+#### res_pjsip_registrar: Include source IP and port in log messages.
+ Author: Joshua C. Colp
+ Date: 2021-03-03
+
+ When registering it can be useful to see the source IP address and
+ port in cases where multiple devices are using the same endpoint
+ or when anonymous is in use.
+
+ ASTERISK-29325
+
+
+#### asterisk: Update copyright.
+ Author: Joshua C. Colp
+ Date: 2021-03-03
+
+ ASTERISK-29326
+
+
+#### AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite.
+ Author: Ben Ford
+ Date: 2021-02-25
+
+ When Asterisk sends a reinvite negotiating T38 faxing, it's possible a
+ crash can occur if the response contains a m=image and zero port. The
+ reinvite callback code now checks session_media to see if it is null or
+ not before trying to access the udptl variable on it.
+
+ ASTERISK-29305
+
+
+#### res_format_attr_h263: Generate valid SDP fmtp for H.263+.
+ Author: Alexander Traud
+ Date: 2021-01-28
+
+ Fixed:
+ * RFC 4629 does not allow the value "0" for MPI, K, and N.
+ * Allow value "0" for PAR.
+ * BPP is printed only when specified because "0" has a meaning.
+
+ New:
+ * Added CPCF and MaxBR.
+ * Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1
+ Although a violation of RFC 3555 section 3, we can support that.
+
+ Changed:
+ * Resorts the CIFs from large to small which partly fixes ASTERISK~29267.
+
+
+#### res_pjsip_nat: Don't rewrite Contact on REGISTER responses.
+ Author: Joshua C. Colp
+ Date: 2021-02-24
+
+ When sending a SIP response to an incoming REGISTER request
+ we don't want to change the Contact header as it will
+ contain the Contacts registered to the AOR and not our own
+ Contact URI.
+
+ ASTERISK-29235
+
+
+#### channel: Fix memory leak in suppress API.
+ Author: Joshua C. Colp
+ Date: 2021-03-03
+
+ A frame suppression API exists as part of channels
+ which allows audio frames to or from a channel to
+ be dropped. The MuteAudio AMI action uses this
+ API to perform its job.
+
+ This API uses a framehook to intercept flowing
+ audio and drop it when appropriate. It is the
+ responsibility of the framehook to free the
+ frame it is given if it changes the frame. The
+ suppression API failed to do this resulting in
+ a leak of audio frames.
+
+ This change adds the freeing of these frames.
+
+ ASTERISK-29071
+
+
+#### res_rtp_asterisk: Check remote ICE reset and reset local ice attrb
+ Author: Salah Ahmed
+ Date: 2021-01-27
+
+ This change will check is the remote ICE session got reset or not by
+ checking the offered ufrag and password with session. If the remote ICE
+ reset session then Asterisk reset its local ufrag and password to reject
+ binding request with Old ufrag and Password.
+
+ ASTERISK-29266
+
+
+#### pjsip: Generate progress (once) when receiving a 180 with a SDP
+ Author: Holger Hans Peter Freyther
+ Date: 2021-01-07
+
+ ASTERISK-29105
+
+
+#### main: With Dutch language year after 2020 is not spoken in say.c
+ Author: Nico Kooijman
+ Date: 2021-02-28
+
+ Implemented the english way of saying the year in ast_say_date_with_format_nl.
+ Currently the numbers are spoken correctly until 2020 and stopped working
+ this year.
+
+ ASTERISK-29297 #close
+ Reported-by: Jacek Konieczny
+
+
+#### res_pjsip: dont return early from registration if init auth fails
+ Author: Nick French
+ Date: 2021-02-24
+
+ If set_outbound_initial_authentication_credentials() fails,
+ handle_client_registration() bails early without creating or
+ sending a register message.
+
+ [set_outbound_initial_authentication_credentials() failures
+ can occur during the process of retrieving an oauth access
+ token.]
+
+ The return from handle_client_registration is ignored, so
+ returning an error doesn't do any good.
+
+ This is a real problem when the registration request is a
+ re-register, because then the registration will still be
+ marked 'active' despite the re-register never being sent at all.
+
+ So instead, log a warning but let the registration be created
+ and sent (and probably fail) and follow the normal registration
+ failed retry/abort logic.
+
+ ASTERISK-29315 #close
+
+
+#### res_fax: validate the remote/local Station ID for UTF-8 format
+ Author: Alexei Gradinari
+ Date: 2021-02-23
+
+ If the remote Station ID contains invalid UTF-8 characters
+ the asterisk fails to publish the Stasis and ReceiveFax status messages.
+
+ json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
+ 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28]
+ 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc]
+ 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x2b) [0x57aa0b]
+ 3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x121) [0x530641]
+ 4: /usr/lib64/asterisk/modules/res_fax.so(+0x44fe) [0x7f27f4bff4fe]
+ ...
+ stasis_channels.c: Error creating message
+
+ json.c: Error building JSON from '{s: s, s: s, s: s, s: s, s: s, s: s, s: o}': Invalid UTF-8 string.
+ 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28]
+ 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc]
+ 2: /usr/lib64/asterisk/modules/res_fax.so(+0x5acd) [0x7f27f4c00acd]
+ ...
+ res_fax.c: Error publishing ReceiveFax status message
+
+ This patch replaces the invalid UTF-8 Station IDs with an empty string.
+
+ ASTERISK-29312 #close
+
+
+#### app_page.c: Don't fail to Page if beep sound file is missing
+ Author: Sean Bright
+ Date: 2021-02-25
+
+ ASTERISK-16799 #close
+
+
+#### res_pjsip_refer: Refactor progress locking and serialization
+ Author: George Joseph
+ Date: 2021-02-19
+
+ Although refer_progress_notify() always runs in the progress
+ serializer, the pjproject evsub module itself can cause the
+ subscription to be destroyed which then triggers
+ refer_progress_on_evsub_state() to clean it up. In this case,
+ it's possible that refer_progress_notify() could get the
+ subscription pulled out from under it while it's trying to use
+ it.
+
+ At one point we tried to have refer_progress_on_evsub_state()
+ push the cleanup to the serializer and wait for its return before
+ returning to pjproject but since pjproject calls its state
+ callbacks with the dialog locked, this required us to unlock the
+ dialog while waiting for the serialized cleanup, then lock it
+ again before returning to pjproject. There were also still some
+ cases where other callers of refer_progress_notify() weren't
+ using the serializer and crashes were resulting.
+
+ Although all callers of refer_progress_notify() now use the
+ progress serializer, we decided to simplify the locking so we
+ didn't have to unlock and relock the dialog in
+ refer_progress_on_evsub_state().
+
+ Now, refer_progress_notify() holds the dialog lock for its
+ duration and since pjproject also holds the dialog lock while
+ calling refer_progress_on_evsub_state() (which does the cleanup),
+ there should be no more chances for the subscription to be
+ cleaned up while still being used to send NOTIFYs.
+
+ To be extra safe, we also now increment the session count on
+ the dialog when we create a progress object and decrement
+ the count when the progress is destroyed.
+
+ ASTERISK-29313
+
+
+#### res_rtp_asterisk: Add packet subtype during RTCP debug when relevant
+ Author: Kevin Harwell
+ Date: 2021-02-24
+
+ For some RTCP packet types the report count is actually the packet's subtype.
+ This was not being reflected in the packet debug output.
+
+ This patch makes it so for some RTCP packet types a "Packet Subtype" is
+ now output in the debug replacing the "Reception reports" (i.e count).
+
+
+#### res_pjsip_session: Always produce offer on re-INVITE without SDP.
+ Author: Joshua C. Colp
+ Date: 2021-02-16
+
+ When PJSIP receives a re-INVITE without an SDP offer the INVITE
+ session library will first call the on_create_offer callback and
+ if unavailable then use the active negotiated SDP as the offer.
+
+ In some cases this would result in a different SDP then was
+ previously used without an incremented SDP version number. The two
+ known cases are:
+
+ 1. Sending an initial INVITE with a set of codecs and having the
+ remote side answer with a subset. The active negotiated SDP would
+ have the pruned list but would not have an incremented SDP version
+ number.
+
+ 2. Using re-INVITE for unhold. We would modify the active negotiated
+ SDP but would not increment the SDP version.
+
+ To solve these, and potential other unknown cases, the on_create_offer
+ callback has now been implemented which produces a fresh offer with
+ incremented SDP version number. This better fits within the model
+ provided by the INVITE session library.
+
+ ASTERISK-28452
+
+
+#### res_odbc_transaction: correctly initialise forcecommit value from DSN.
+ Author: Jaco Kroon
+ Date: 2021-02-23
+
+ Also improve the in-process documentation to clarify that the value is
+ initialised from the DSN and not default false, but that the DSN's value
+ is default false if unset.
+
+ ASTERISK-29311 #close
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### res_pjsip_session.c: Check topology on re-invite.
+ Author: Ben Ford
+ Date: 2021-02-15
+
+ Removes an unnecessary check for the conditional that compares the
+ stream topologies to see if they are equal to suppress re-invites. This
+ was a problem when a Digium phone received an INVITE that offered codecs
+ different than what it supported, causing Asterisk to send the
+ re-invite.
+
+ ASTERISK-29303
+
+
+#### res_config_pgsql: Limit realtime_pgsql() to return one (no more) record.
+ Author: Boris P. Korzun
+ Date: 2021-02-15
+
+ Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function.
+
+ ASTERISK-29293 #close
+
+
+#### app_queue: Fix conversion of complex extension states into device states
+ Author: Ivan Poddubnyi
+ Date: 2019-09-13
+
+ Queue members using dialplan hints as a state interface must handle
+ INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE.
+
+ ASTERISK-28369
+
+
+#### app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS
+ Author: Jaco Kroon
+ Date: 2021-02-10
+
+ This partially reverts commit 3d1bf3c537bba0416f691f48165fdd0a32554e8a,
+ specifically for app.h.
+
+ This works with both gcc 9.3.0 and 10.2.0 now, both for C and C++ (as
+ tested with external modules).
+
+ ASTERISK-29287
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### chan_sip: Filter pass-through audio/video formats away, again.
+ Author: Alexander Traud
+ Date: 2021-02-05
+
+ Instead of looking for pass-through formats in the list of transcodable
+ formats (which is going to find nothing), go through the result which
+ is going to be the jointcaps of the tech_pvt of the channel. Finally,
+ only with that list, ast_format_cap_remove(.) is going to succeed.
+
+ This restores the behaviour of Asterisk 1.8. However, it does not fix
+ ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
+ Here, only chan_sip is fixed because PJSIP does not even call
+ ast_rtp_instance_available_formats -> ast_translate_available_format.
+
+
+#### func_odbc: Introduce minargs config and expose ARGC in addition to ARGn.
+ Author: Jaco Kroon
+ Date: 2021-02-17
+
+ minargs enables enforcing of minimum count of arguments to pass to
+ func_odbc, so if you're unconditionally using ARG1 through ARG4 then
+ this should be set to 4. func_odbc will generate an error in this case,
+ so for example
+
+ [FOO]
+ minargs = 4
+
+ and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a
+ potentially leaked ARG4 from Gosub().
+
+ ARGC is needed if you're using optional argument, to verify whether or
+ not an argument has been passed, else it's possible to use a leaked ARGn
+ from Gosub (app_stack). So now you can safely do
+ ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute.
+ Author: Sebastien Duthil
+ Date: 2021-01-13
+
+ ASTERISK-29244
+
+
+#### AST-2021-002: Remote crash possible when negotiating T.38
+ Author: Kevin Harwell
+ Date: 2021-02-01
+
+ When an endpoint requests to re-negotiate for fax and the incoming
+ re-invite is received prior to Asterisk sending out the 200 OK for
+ the initial invite the re-invite gets delayed. When Asterisk does
+ finally send the re-inivite the SDP includes streams for both audio
+ and T.38.
+
+ This happens because when the pending topology and active topologies
+ differ (pending stream is not in the active) in the delayed scenario
+ the pending stream is appended to the active topology. However, in
+ the fax case the pending stream should replace the active.
+
+ This patch makes it so when a delay occurs during fax negotiation,
+ to or from, the audio stream is replaced by the T.38 stream, or vice
+ versa instead of being appended.
+
+ Further when Asterisk sent the re-invite with both audio and T.38,
+ and the endpoint responded with a declined T.38 stream then Asterisk
+ would crash when attempting to change the T.38 state.
+
+ This patch also puts in a check that ensures the media state has a
+ valid fax session (associated udptl object) before changing the
+ T.38 state internally.
+
+ ASTERISK-29203 #close
+
+
+#### rtp: Enable srtp replay protection
+ Author: Alexander Traud
+ Date: 2021-01-26
+
+ Add option "srtpreplayprotection" rtp.conf to enable srtp
+ replay protection.
+
+ ASTERISK-29260
+ Reported by: Alexander Traud
+
+
+#### res_pjsip_diversion: Fix adding more than one histinfo to Supported
+ Author: Ivan Poddubnyi
+ Date: 2020-12-28
+
+ New responses sent within a PJSIP sessions are based on those that were
+ sent before. Therefore, adding/modifying a header once causes it to be
+ sent on all responses that follow.
+
+ Sending 181 Call Is Being Forwarded many times first adds "histinfo"
+ duplicated more and more, and eventually overflows past the array
+ boundary.
+
+ This commit adds a check preventing adding "histinfo" more than once,
+ and skipping it if there is no more space in the header.
+
+ Similar overflow situations can also occur in res_pjsip_path and
+ res_pjsip_outbound_registration so those were also modified to
+ check the bounds and suppress duplicate Supported values.
+
+ ASTERISK-29227
+ Reported by: Ivan Poddubny
+
+
+#### res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
+ Author: Sean Bright
+ Date: 2020-12-11
+
+ ASTERISK-29205 #close
+
+
+#### pjsip: Make modify_local_offer2 tolerate previous failed SDP.
+ Author: Joshua C. Colp
+ Date: 2021-02-05
+
+ If a remote side is broken and sends an SDP that can not be
+ negotiated the call will be torn down but there is a window
+ where a second 183 Session Progress or 200 OK that is forked
+ can be received that also attempts to negotiate SDP. Since
+ the code marked the SDP negotiation as being done and complete
+ prior to this it assumes that there is an active local and remote
+ SDP which it can modify, while in fact there is not as the SDP
+ did not successfully negotiate. Since there is no local or remote
+ SDP a crash occurs.
+
+ This patch changes the pjmedia_sdp_neg_modify_local_offer2
+ function to no longer assume that a previous SDP negotiation
+ was successful.
+
+ ASTERISK-29196
+
+
+#### res_pjsip_refer: Always serialize calls to refer_progress_notify
+ Author: George Joseph
+ Date: 2021-02-09
+
+ refer_progress_notify wasn't always being called from the progress
+ serializer. This could allow clearing notification->progress->sub
+ in one thread while another was trying to use it.
+
+ * Instances where refer_progress_notify was being called in-line,
+ have been changed to use ast_sip_push_task().
+
+
+#### core_unreal: Fix T.38 faxing when using local channels.
+ Author: Ben Ford
+ Date: 2021-01-11
+
+ After some changes to streams and topologies, receiving fax through
+ local channels stopped working. This change adds a stream topology with
+ a stream of type IMAGE to the local channel pair and allows fax to be
+ received.
+
+ ASTERISK-29035 #close
+
+
+#### format_wav: Support of MIME-type for wav16
+ Author: Boris P. Korzun
+ Date: 2021-02-02
+
+ Provided a support of a MIME-type for wav16. Added new MIME-type
+ for classic wav.
+
+ ASTERISK-29275 #close
+
+
+#### chan_sip: Allow [peer] without audio (text+video).
+ Author: Alexander Traud
+ Date: 2021-02-05
+
+ Two previous commits, 620d9f4 and 6d980de, allow to set up a call
+ without audio, again. That was introduced originally with commit f04d5fb
+ but changed and broke over time. The original commit missed one
+ scenario: A [peer] section in sip.conf, which does not allow audio at
+ all. In that case, chan_sip rejected the call, although even when the
+ requester offered no audio. Now, chan_sip does not check whether there
+ is no audio format but checks whether there is no format in general. In
+ other words, if there is at least one format to offer, the call succeeds.
+
+ However, to prevent calls with no-audio, chan_sip still rejects calls
+ when both call parties (caller = requester of the call *and* callee =
+ [peer] section in sip.conf) included audio. In such a case, it is
+ expected that the call should have audio.
+
+ ASTERISK-29280
+
+
+#### chan_iax2.c: Require secret and auth method if encryption is enabled
+ Author: George Joseph
+ Date: 2021-01-28
+
+ If there's no secret specified for an iax2 peer and there's no secret
+ specified in the dial string, Asterisk will crash if the auth method
+ requested by the peer is MD5 or plaintext. You also couldn't specify
+ a default auth method in the [general] section of iax.conf so if you
+ don't have static peers defined and just use the dial string, Asterisk
+ will still crash even if you have a secret specified in the dial string.
+
+ * Added logic to iax2_call() and authenticate_reply() to print
+ a warning and hanhup the call if encryption is requested and
+ there's no secret or auth method. This prevents the crash.
+
+ * Added the ability to specify a default "auth" in the [general]
+ section of iax.conf.
+
+ ASTERISK-29624
+ Reported by: N A
+
+
+#### app_read: Release tone zone reference on early return.
+ Author: Sean Bright
+ Date: 2021-02-03
+
+
+#### chan_sip: Set up calls without audio (text+video), again.
+ Author: Alexander Traud
+ Date: 2021-01-27
+
+ The previous commit 6d980de fixed this issue in the core of Asterisk.
+ With that, each channel technology can be used without audio
+ theoretically. Practically, the channel-technology driver chan_sip
+ turned out to have an invalid check preventing that. chan_sip tested
+ whether there is at least one audio format. However, chan_sip has to
+ test whether there is at least one format. More cannot be tested while
+ requesting chan_sip because only the [general] capabilities but not the
+ [peer] caps are known yet. And the [peer] caps might not be a subset or
+ show any intersection with the [general] caps. This change here fixes
+ this.
+
+ The original commit f04d5fb, thirteen years ago, contained a software
+ bug as it passed ANY audio capability to the channel-technology driver.
+ Instead, it should have passed NO audio format. Therefore, this
+ addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8.
+ Then, Asterisk 10 changed that from ANY to NO, but nobody reported since
+ then.
+
+ ASTERISK-29265
+
+
+#### chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable
+ Author: Dan Cropp
+ Date: 2021-01-22
+
+ When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is
+ 0 when no protocl specific error
+ SIP example of failure, 3xx-6xx for the SIP error code received
+
+ This allows applications to perform actions based on the failure
+ reason.
+
+ ASTERISK-29252 #close
+ Reported-by: Dan Cropp
+
+
+#### channel: Set up calls without audio (text+video), again.
+ Author: Alexander Traud
+ Date: 2021-01-22
+
+ ASTERISK-29259
+
+
+#### res/res_pjsip.c: allow user=phone when number contain *#
+ Author: roadkill
+ Date: 2021-01-22
+
+ if From number contain * or # asterisk will not add user=phone
+
+ Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY
+ this is a problem when you want to send call to ISUP
+ as they will disregard the From header and either replace From with anonymous or with p-asserted-identity
+
+ ASTERISK-29261
+ Reported by: Mark Petersen
+ Tested by: Mark Petersen
+
+
+#### chan_sip: SDP: Reject audio streams correctly.
+ Author: Alexander Traud
+ Date: 2021-01-21
+
+ This completes the fix for ASTERISK_24543. Only when the call is an
+ outgoing call, consult and append the configured format capabilities
+ (p->caps). When all audio formats got rejected the negotiated format
+ capabilities (p->jointcaps) contain no audio formats for incoming
+ calls. This is required when there are other accepted media streams.
+
+ ASTERISK-29258
+
+
+#### main/frame: Add missing control frame names to ast_frame_subclass2str
+ Author: Ivan Poddubnyi
+ Date: 2021-01-22
+
+ Log proper control frame names instead of "Unknown control '14'", etc.
+
+
+#### res_musiconhold: Add support of various URL-schemes by MoH.
+ Author: Boris P. Korzun
+ Date: 2021-01-23
+
+ Provided a support of variuos URL-schemes for res_musiconhold,
+ registered by ast_bucket_scheme_register().
+
+ ASTERISK-29262 #close
+
+
+#### AC_HEADER_STDC causes a compile failure with autoconf 2.70
+ Author: Jaco Kroon
+ Date: 2021-01-08
+
+ From https://www.mail-archive.com/bug-autoconf@gnu.org/msg04408.html
+
+ > ... the long-obsolete AC_HEADER_STDC, previously used internally by
+ > AC_INCLUDES_DEFAULT, used AC_EGREP_HEADER. The AC_HEADER_STDC macro
+ > is now a no-op (and is not used at all within Autoconf anymore), so
+ > that change is likely what made the first use of AC_EGREP_HEADER the
+ > one inside the if condition, causing the observed results.
+
+ The implication is that the test does nothing anyway, and due to it
+ being a no-op from 2.70 onwards, results in the required not being set
+ to yes, resulting in ./configure to fail.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang.
+ Author: Alexander Traud
+ Date: 2021-01-15
+
+ Otherwise, Clang 10 warned because of logical-not-parentheses.
+
+
+#### res_pjsip_session: Avoid sometimes-uninitialized warning with Clang.
+ Author: Alexander Traud
+ Date: 2021-01-15
+
+ ASTERISK-29248
+
+
+#### res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet
+ Author: Sean Bright
+ Date: 2021-01-14
+
+ The last argument to ast_copy_string() is the buffer size, not the
+ number of characters, so we add 1 to avoid stamping out the final \n
+ in the persisted SUBSCRIBE message.
+
+
+#### chan_pjsip.c: Add parameters to frame in indicate.
+ Author: Ben Ford
+ Date: 2021-01-11
+
+ There are a couple of parameters (datalen and data) that do not get set
+ in chan_pjsip_indicate which could cause an Invalid message to pop up
+ for things such as fax. This patch adds them to the frame.
+
+
+#### res/res_pjsip_session.c: Check that media type matches in function ast_sip_ses..
+ Author: Robert Cripps
+ Date: 2020-12-22
+
+ Check ast_media_type matches when a ast_sip_session_media is found
+ otherwise when transitioning from say image to audio, the wrong
+ session is returned in the first if statement.
+
+ ASTERISK-29220 #close
+
+
+#### Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
+ Author: Jean Aunis
+ Date: 2020-12-30
+
+ When both a tech subscription and an endpoint subscription exist for a given
+ endpoint, TextMessageReceived events are dispatched to the tech subscription
+ only.
+
+ ASTERISK-29229
+
+
+#### chan_sip: SDP: Sidestep stream parsing when its media is disabled.
+ Author: Alexander Traud
+ Date: 2020-12-23
+
+ Previously, chan_sip parsed all known media streams in an SDP offer
+ like video (and text) even when videosupport=no (and textsupport=no).
+ This wasted processor power. Furthermore, chan_sip accepted SDP offers,
+ including no audio but just video (or text) streams although
+ videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
+ offer instead of individual streams when they had encryption (SDES-sRTP)
+ unexpectedly enabled.
+
+ ASTERISK-29238
+ ASTERISK-29237
+ ASTERISK-29222
+
+
+#### chan_pjsip: Assign SIPDOMAIN after creating a channel
+ Author: Ivan Poddubnyi
+ Date: 2020-12-29
+
+ session->channel doesn't exist until chan_pjsip creates it, so intead of
+ setting a channel variable every new incoming call sets one and the same
+ global variable.
+
+ This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
+ a newly created channel, it also removes a misleading reference to
+ channel->session used to fetch call pickup configuraion.
+
+ ASTERISK-29240
+
+
+#### chan_pjsip: Stop queueing control frames twice on outgoing channels
+ Author: Ivan Poddubnyi
+ Date: 2020-12-31
+
+ The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
+ This resulted in extra noise in logs (for example, "is making progress"
+ and "is ringing" get logged twice by app_dial), as well as in noise in
+ signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.
+
+ This change splits the response handler into 2 functions:
+ - one for updating HANGUPCAUSE, which is still called twice,
+ - another that does the rest, which is called only once as before.
+
+ ASTERISK-28016
+ Reported-by: Alex Hermann
+
+ ASTERISK-28549
+ Reported-by: Gant Liu
+
+ ASTERISK-28185
+ Reported-by: Julien
+
+
+#### contrib/systemd: Added note on common issues with systemd and asterisk
+ Author: Jaco Kroon
+ Date: 2020-12-18
+
+ With newer version of linux /var/run/ is a symlink to /run/ that has
+ been turned into tmpfs.
+
+ Added note that if asterisk has to bind to a specific IP that
+ systemd has to wait until the network is up.
+
+ Added note on how to make sure that the environment variable
+ HOSTNAME is included.
+
+ ASTERISK-29216
+ Reported by: Mark Petersen
+ Tested by: Mark Petersen
+
+
+#### Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other st..
+ Author: George Joseph
+ Date: 2021-01-07
+
+ This reverts commit 2fe76dd816706f045ecbc44bf8ad6498977415b3.
+
+ Reason for revert: Too many issues reported. Need to research and correct.
+
+ ASTERISK-29230
+ ASTERISK-29231
+ Reported by: Michael Maier
+
+
+#### func_lock: fix multiple-channel-grant problems.
+ Author: Jaco Kroon
+ Date: 2020-12-18
+
+ Under contention it becomes possible that multiple channels will be told
+ they successfully obtained the lock, which is a bug. Please refer
+
+ ASTERISK-29217
+
+ This introduces a couple of changes.
+
+ 1. Replaces requesters ao2 container with simple counter (we don't
+ really care who is waiting for the lock, only how many). This is
+ updated undex ->mutex to prevent memory access races.
+ 2. Correct semantics for ast_cond_timedwait() as described in
+ pthread_cond_broadcast(3P) is used (multiple threads can be released
+ on a single _signal()).
+ 3. Module unload races are taken care of and memory properly cleaned
+ up.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### pbx_lua: Add LUA_VERSIONS environment variable to ./configure.
+ Author: Jaco Kroon
+ Date: 2020-12-23
+
+ On Gentoo it's possible to have multiple lua versions installed, all
+ with a path of /usr, so it's not possible to use the current --with-lua
+ option to determisticly pin to a specific version as is required by the
+ Gentoo PMS standards.
+
+ This environment variable allows to lock to specific versions,
+ unversioned check will be skipped if this variable is supplied.
+
+ Signed-off-by: Jaco Kroon <jaco@uls.co.za>
+
+#### app_mixmonitor: cleanup datastore when monitor thread fails to launch
+ Author: Kevin Harwell
+ Date: 2020-12-23
+
+ launch_monitor_thread is responsible for creating and initializing
+ the mixmonitor, and dependent data structures. There was one off
+ nominal path after the datastore gets created that triggers when
+ the channel being monitored is hung up prior to monitor starting
+ itself.
+
+ If this happened the monitor thread would not "launch", and the
+ mixmonitor object and associated objects are freed, including the
+ underlying datastore data object. However, the datastore itself was
+ not removed from the channel, so when the channel eventually gets
+ destroyed it tries to access the previously freed datastore data
+ and crashes.
+
+ This patch removes and frees datastore object itself from the channel
+ before freeing the mixmonitor object thus ensuring the channel does
+ not call it when destroyed.
+
+ ASTERISK-28947 #close
+
+
+#### app_voicemail: Prevent deadlocks when out of ODBC database connections
+ Author: Sean Bright
+ Date: 2020-12-24
+
+ ASTERISK-28992 #close
+
+
+#### chan_pjsip: Incorporate channel reference count into transfer_refer().
+ Author: Dan Cropp
+ Date: 2020-12-07
+
+ Add channel reference count for PJSIP REFER. The call could be terminated
+ prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
+ occurred several minutes later, it would attempt to access a session which was
+ no longer valid. Terminate event subscription if pjsip_xfer_initiate() or
+ pjsip_xfer_send_request() fails in transfer_refer().
+
+ ASTERISK-29201 #close
+ Reported-by: Dan Cropp
+
+
+#### pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type
+ Author: Kevin Harwell
+ Date: 2020-12-22
+
+ A prior patch segmented channel snapshots, and changed the underlying
+ data object type associated with ast_channel_snapshot_type stasis
+ messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but
+ now it type ast_channel_snapshot_update.
+
+ When publishing ast_channel_snapshot_type in pbx_realtime the
+ ast_channel_snapshot was being passed in as the message data
+ object. When a handler, expecting a data object type of
+ ast_channel_snapshot_update, dereferenced this value a crash
+ would occur.
+
+ This patch makes it so pbx_realtime now uses the expected type, and
+ channel snapshot publish method when publishing.
+
+ ASTERISK-29168 #close
+
+
+#### asterisk: Export additional manager functions
+ Author: Sean Bright
+ Date: 2020-12-18
+
+ Rename check_manager_enabled() and check_webmanager_enabled() to begin
+ with ast_ so that the symbols are automatically exported by the
+ linker.
+
+ ASTERISK~29184
+
+
+#### res_pjsip: Prevent segfault in UDP registration with flow transports
+ Author: Nick French
+ Date: 2020-12-19
+
+ Segfault occurs during outbound UDP registration when all
+ transport states are being iterated over. The transport object
+ in the transport is accessed, but flow transports have a NULL
+ transport object.
+
+ Modify to not iterate over any flow transport
+
+ ASTERISK-29210 #close
+
+
+#### codecs: Remove test-law.
+ Author: Alexander Traud
+ Date: 2020-12-01
+
+ This was dead code, test code introduced with Asterisk 13. This was
+ found while analyzing ASTERISK_28416 and ASTERISK_29185. This change
+ partly fixes, not closes those two issues.
+
+
+#### res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
+ Author: Torrey Searle
+ Date: 2020-12-22
+
+ Add a check to see if the URI is a Tel URI and prevent crashing on
+ trying to retrieve the reason parameter.
+
+ ASTERISK-29191
+ ASTERISK-29219
+
+
+#### chan_vpb.cc: Fix compile errors.
+ Author: Richard Mudgett
+ Date: 2020-12-26
+
+ Fix the usual compile problem when someone adds a new callback to struct
+ ast_channel_tech.
+
+
+#### res_pjsip_session.c: Fix compiler warnings.
+ Author: Richard Mudgett
+ Date: 2020-12-26
+
+ AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an
+ unsigned long on all machines.
+
+
+#### res_pjsip_session: Fixed NULL active media topology handle
+ Author: Sungtae Kim
+ Date: 2020-12-13
+
+ Added NULL pointer check to prevent Asterisk crash.
+
+ ASTERISK-29215
+
+
+#### app_chanspy: Spyee information missing in ChanSpyStop AMI Event
+ Author: Sean Bright
+ Date: 2020-12-11
+
+ The documentation in the wiki says there should be spyee-channel
+ information elements in the ChanSpyStop AMI event.
+
+ https://wiki.asterisk.org/wiki/x/Xc5uAg
+
+ However, this is not the case in Asterisk <= 16.10.0 Version. We're
+ using these Spyee* arguments since Asterisk 11.x, so these arguments
+ vanished in Asterisk 12 or higher.
+
+ For maximum compatibility, we still send the ChanSpyStop event even if
+ we are not able to find any 'Spyee' information.
+
+ ASTERISK-28883 #close
+
+
+#### res_ari: Fix wrong media uri handle for channel play
+ Author: Sungtae Kim
+ Date: 2020-12-01
+
+ Fixed wrong null object handle in
+ /channels/<channel_id>/play request handler.
+
+ ASTERISK-29188
+
+
+#### logger.c: Automatically add a newline to formats that don't have one
+ Author: George Joseph
+ Date: 2020-12-10
+
+ Scope tracing allows you to not specify a format string or variable,
+ in which case it just prints the indent, file, function, and line
+ number. The trace output automatically adds a newline to the end
+ in this case. If you also have debugging turned on for the module,
+ a debug message is also printed but the standard log functionality
+ which prints it doesn't add the newline so you have messages
+ that don't break correctly.
+
+ * format_log_message_ap(), which is the common log
+ message formatter for all channels, now adds a
+ newline to the end of format strings that don't
+ already have a newline.
+
+ ASTERISK-29209
+ Reported by: Alexander Traud
+
+
+#### res_pjsip_nat.c: Create deep copies of strings when appropriate
+ Author: Pirmin Walthert
+ Date: 2020-12-08
+
+ In rewrite_uri asterisk was not making deep copies of strings when
+ changing the uri. This was in some cases causing garbage in the route
+ header and in other cases even crashing asterisk when receiving a
+ message with a record-route header set. Thanks to Ralf Kubis for
+ pointing out why this happens. A similar problem was found in
+ res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
+ to avoid garbage in CANCEL messages.
+
+ ASTERISK-29024 #close
+
+
+#### res_musiconhold: Don't crash when real-time doesn't return any entries
+ Author: Nathan Bruning
+ Date: 2020-12-11
+
+ ASTERISK-29211 #close
+
+
+#### res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
+ Author: Joshua C. Colp
+ Date: 2020-12-16
+
+ This adds support for both Digium and Sangoma user agent strings
+ for the Sangoma specific body supplement.
+
+
+#### pjsip: Match lifetime of INVITE session to our session.
+ Author: Joshua C. Colp
+ Date: 2020-10-29
+
+ In some circumstances it was possible for an INVITE
+ session to be destroyed while we were still using it.
+ This occurred due to the reference on the INVITE session
+ being released internally as a result of its state
+ changing to DISCONNECTED.
+
+ This change adds a reference to the INVITE session
+ which is released when our own session is destroyed,
+ ensuring that the INVITE session remains valid for
+ the lifetime of our session.
+
+ ASTERISK-29022
+
+
+#### res_http_media_cache.c: Set reasonable number of redirects
+ Author: Sean Bright
+ Date: 2020-11-21
+
+ By default libcurl does not follow redirects, so we explicitly enable
+ it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl
+ will follow up to CURLOPT_MAXREDIRS redirects, which by default is
+ configured to be unlimited.
+
+ This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If
+ we determine at some point that this needs to be increased on
+ configurable it is a trivial change.
+
+ ASTERISK-29173 #close
+
+
+#### Introduce astcachedir, to be used for temporary bucket files
+ Author: lvl
+ Date: 2020-10-29
+
+ As described in the issue, /tmp is not a suitable location for a
+ large amount of cached media files, since most distributions make
+ /tmp a RAM-based tmpfs mount with limited capacity.
+
+ I opted for a location that can be configured separately, as opposed
+ to using a subdirectory of spooldir, given the different storage
+ profile (transient files vs files that might stay there indefinitely).
+
+ This commit just makes the cache directory configurable, and changes
+ the default location from /tmp to /var/cache/asterisk.
+
+ ASTERISK-29143
+
+
+#### media_cache: Fix reference leak with bucket file metadata
+ Author: Sean Bright
+ Date: 2020-11-23
+
+
+#### res_pjsip_stir_shaken: Fix module description
+ Author: Stanislav
+ Date: 2020-11-24
+
+ the 'J' is missing in module description.
+ "PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk"
+
+ ASTERISK-29175 #close
+
+
+#### voicemail: add option 'e' to play greetings as early media
+ Author: Joshua C. Colp
+ Date: 2020-10-12
+
+ When using this option, answering the channel is deferred until
+ all prompts/greetings have been played and the caller is about
+ to leave their message.
+
+ ASTERISK-29118 #close
+
+
+#### loader: Sync load- and build-time deps.
+ Author: Alexander Traud
+ Date: 2020-11-02
+
+ In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO.
+
+ ASTERISK-29148
+
+
+#### CHANGES: Remove already applied CHANGES update
+ Author: Sean Bright
+ Date: 2020-11-18
+
+
+#### res_pjsip: set Accept-Encoding to identity in OPTIONS response
+ Author: Alexander Greiner-Baer
+ Date: 2020-11-17
+
+ RFC 3261 says that the Accept-Encoding header should be present
+ in an options response. Permitted values according to RFC 2616
+ are only compression algorithms like gzip or the default identity
+ encoding. Therefore "text/plain" is not a correct value here.
+ As long as the header is hard coded, it should be set to "identity".
+
+ Without this fix an Alcatel OmniPCX periodically logs warnings like
+ "[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed"
+ on a SIP Trunk.
+
+ ASTERISK-29165 #close
+
+
+#### chan_sip: Remove unused sip_socket->port.
+ Author: Alexander Traud
+ Date: 2020-11-04
+
+ 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
+ vanished. However, the struct member itself and all seven set/uses
+ remained as dead code.
+
+ ASTERISK-28798
+
+
+#### bridge_basic: Fixed setup of recall channels
+ Author: Boris P. Korzun
+ Date: 2020-11-13
+
+ Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641.
+ common_recall_channel_setup() setups common things on the recalled transfer
+ target, but used same target as source instead trasfered.
+
+ ASTERISK-29161 #close
+
+
+#### modules.conf: Align the comments for more conclusiveness.
+ Author: Alexander Traud
+ Date: 2020-11-03
+
+
+#### app_queue: Fix deadlock between update and show queues
+ Author: George Joseph
+ Date: 2020-11-11
+
+ Operations that update queues when shared_lastcall is set lock the
+ queue in question, then have to lock the queues container to find the
+ other queues with the same member. On the other hand, __queues_show
+ (which is called by both the CLI and AMI) does the reverse. It locks
+ the queues container, then iterates over the queues locking each in
+ turn to display them. This creates a deadlock.
+
+ * Moved queue print logic from __queues_show to a separate function
+ that can be called for a single queue.
+
+ * Updated __queues_show so it doesn't need to lock or traverse
+ the queues container to show a single queue.
+
+ * Updated __queues_show to snap a copy of the queues container and iterate
+ over that instead of locking the queues container and iterating over
+ it while locked. This prevents us from having to hold both the
+ container lock and the queue locks at the same time. This also
+ allows us to sort the queue entries.
+
+ ASTERISK-29155
+
+
+#### res_pjsip_outbound_registration.c: Use our own scheduler and other stuff
+ Author: George Joseph
+ Date: 2020-11-02
+
+ * Instead of using the pjproject timer heap, we now use our own
+ pjsip_scheduler. This allows us to more easily debug and allows us to
+ see times in "pjsip show/list registrations" as well as being able to
+ see the registrations in "pjsip show scheduled_tasks".
+
+ * Added the last registration time, registration interval, and the next
+ registration time to the CLI output.
+
+ * Removed calls to pjsip_regc_info() except where absolutely necessary.
+ Most of the calls were just to get the server and client URIs for log
+ messages so we now just save them on the client_state object when we
+ create it.
+
+ * Added log messages where needed and updated most of the existong ones
+ to include the registration object name at the start of the message.
+
+
+#### pjsip_scheduler.c: Add type ONESHOT and enhance cli show command
+ Author: George Joseph
+ Date: 2020-11-02
+
+ * Added a ONESHOT type that never reschedules.
+
+ * Added "like" capability to "pjsip show scheduled_tasks" so you can do
+ the following:
+
+ CLI> pjsip show scheduled_tasks like outreg
+ PJSIP Scheduled Tasks:
+
+ Task Name Interval Times Run ...
+ ============================================= ========= ========= ...
+ pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ...
+ pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ...
+
+ * Fixed incorrect display of "Next Start".
+
+ * Compacted the displays of times in the CLI.
+
+ * Added two new functions (ast_sip_sched_task_get_times2,
+ ast_sip_sched_task_get_times_by_name2) that retrieve the interval,
+ next start time, and next run time in addition to the times already
+ returned by ast_sip_sched_task_get_times().
+
+
+#### sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data
+ Author: Alexei Gradinari
+ Date: 2020-10-02
+
+ The data can be freed if the old object '_data' is the same object as
+ new 'data'. Because at first the object is unreferenced which can lead
+ to destroying it.
+
+ This could happened in res_pjsip_pubsub when the publication is updated
+ which could lead to segfault in function publish_expire.
+
+
+#### res_pjsip/config_transport: Load and run without OpenSSL.
+ Author: Alexander Traud
+ Date: 2020-10-30
+
+ ASTERISK-28933
+ Reported-by: Walter Doekes
+
+
+#### res_stir_shaken: Include OpenSSL headers where used actually.
+ Author: Alexander Traud
+ Date: 2020-10-30
+
+ This avoids the inclusion of the OpenSSL headers in the public header,
+ which avoids one external library dependency in res_pjsip_stir_shaken.
+
+
+#### func_curl.c: Allow user to set what return codes constitute a failure.
+ Author: Dovid Bender
+ Date: 2020-10-18
+
+ Currently any response from res_curl where we get an answer from the
+ web server, regardless of what the response is (404, 403 etc.) Asterisk
+ currently treats it as a success. This patch allows you to set which
+ codes should be considered as a failure by Asterisk. If say we set
+ failurecodes=404,403 then when using curl in realtime if a server gives
+ a 404 error Asterisk will try to failover to the next option set in
+ extconfig.conf
+
+ ASTERISK-28825
+
+ Reported by: Dovid Bender
+ Code by: Gobinda Paul
+
+
+#### AST-2020-001 - res_pjsip: Return dialog locked and referenced
+ Author: Kevin Harwell
+ Date: 2020-11-04
+
+ pjproject returns the dialog locked and with a reference. However,
+ in Asterisk the method that handles this decrements the reference
+ and removes the lock prior to returning. This makes it possible,
+ under some circumstances, for another thread to free said dialog
+ before the thread that created it attempts to use it again. Of
+ course when the thread that created it tries to use a freed dialog
+ a crash can occur.
+
+ This patch makes it so Asterisk now returns the newly created
+ dialog both locked, and with an added reference. This allows the
+ caller to de-reference, and unlock the dialog when it is safe to
+ do so.
+
+ In the case of a new SIP Invite the lock, and reference are now
+ held for the entirety of the new invite handling process.
+ Otherwise it's possible for the dialog, or its dependent objects,
+ like the transaction, to disappear. For example if there is a TCP
+ transport error.
+
+ ASTERISK-29057 #close
+
+
+#### AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit.
+ Author: Ben Ford
+ Date: 2020-11-03
+
+ If Asterisk sends out and INVITE and receives a challenge with a
+ different nonce value each time, it will continually send out INVITEs,
+ even if the call is hung up. The endpoint must be configured for
+ outbound authentication in order for this to occur. A limit has been set
+ on outbound INVITEs so that, once reached, Asterisk will stop sending
+ INVITEs and the transaction will terminate.
+
+ ASTERISK-29013
+
+
+#### sip_to_pjsip.py: Handle #include globs and other fixes
+ Author: Sean Bright
+ Date: 2020-10-29
+
+ * Wildcards in #includes are now properly expanded
+
+ * Implement operators for Section class to allow sorting
+
+ ASTERISK-29142 #close
+
+
+#### Compiler fixes for GCC with -Og
+ Author: Alexander Traud
+ Date: 2020-10-29
+
+ ASTERISK-29144
+
+
+#### Compiler fixes for GCC when printf %s is NULL
+ Author: Alexander Traud
+ Date: 2020-10-30
+
+ ASTERISK-29146
+
+
+#### Compiler fixes for GCC with -Os
+ Author: Alexander Traud
+ Date: 2020-10-29
+
+ ASTERISK-29145
+
+
+#### chan_sip: On authentication, pick MD5 for sure.
+ Author: Alexander Traud
+ Date: 2020-10-23
+
+ RFC 8760 added new digest-access-authentication schemes. Testing
+ revealed that chan_sip does not pick MD5 if several schemes are offered
+ by the User Agent Server (UAS). This change does not implement any of
+ the new schemes like SHA-256. This change makes sure, MD5 is picked so
+ UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can
+ still be used. This should have worked since day one because SIP/2.0
+ already envisioned several schemes (see RFC 3261 and its augmented BNF
+ for 'algorithm' which includes 'token' as third alternative; note: if
+ 'algorithm' was not present, MD5 is still assumed even in RFC 7616).
+
+
+#### main/say: Work around gcc 9 format-truncation false positive
+ Author: Walter Doekes
+ Date: 2020-06-04
+
+ Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0
+ Warning:
+ say.c:2371:24: error: ‘%d’ directive output may be truncated writing
+ between 1 and 11 bytes into a region of size 10
+ [-Werror=format-truncation=]
+ 2371 | snprintf(buf, 10, "%d", num);
+ say.c:2371:23: note: directive argument in the range [-2147483648, 9]
+
+ That's not possible though, as the if() starts out checking for (num < 0),
+ making this Warning a false positive.
+
+ (Also replaced some else<TAB>if with else<SP>if while in the vicinity.)
+
+
+#### res_pjsip, res_pjsip_session: initialize local variables
+ Author: Kevin Harwell
+ Date: 2020-10-19
+
+ This patch initializes a couple of local variables to some default values.
+ Interestingly, in the 'pj_status_t dlg_status' case the value not being
+ initialized caused memory to grow, and not be recovered, in the off nominal
+ path (at least on my machine).
+
+
+#### install_prereq: Add GMime 3.0.
+ Author: Alexander Traud
+ Date: 2020-10-23
+
+ Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not
+ come with GMime 3.0. aptitude ignores any missing package. Therefore,
+ it installs the correct package(s). However, in Ubuntu 18.04 LTS and
+ Ubuntu 20.04 LTS, both versions are installed alongside although only
+ one is really needed.
+
+
+#### BuildSystem: Enable Lua 5.4.
+ Author: Alexander Traud
+ Date: 2020-10-23
+
+ Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested
+ at runtime with pbx_lua. Until then, use the lowest available version
+ of Lua, if you enabled the module pbx_lua at all.
+
+
+#### res_pjsip_session: Restore calls to ast_sip_message_apply_transport()
+ Author: Nick French
+ Date: 2020-10-13
+
+ Commit 44bb0858cb3ea6a8db8b8d1c7fedcfec341ddf66 ("debugging: Add enough
+ to choke a mule") accidentally removed calls to
+ ast_sip_message_apply_transport when it was attempting to just add
+ debugging code.
+
+ The kiss of death was saying that there were no functional changes in
+ the commit comment.
+
+ This makes outbound calls that use the 'flow' transport mechanism fail,
+ since this call is used to relay headers into the outbound INVITE
+ requests.
+
+ ASTERISK-29124 #close
+
+
+#### features.conf.sample: Sample sound files incorrectly quoted
+ Author: Sean Bright
+ Date: 2020-10-22
+
+ ASTERISK-29136 #close
+
+
+#### logger.conf.sample: add missing comment mark
+ Author: Andrew Siplas
+ Date: 2020-10-12
+
+ Add missing comment mark from stock configuration.
+
+ ASTERISK-29123 #close
+
+
+#### res_pjsip: Adjust outgoing offer call pref.
+ Author: Joshua C. Colp
+ Date: 2020-10-06
+
+ This changes the outgoing offer call preference
+ default option to match the behavior of previous
+ versions of Asterisk.
+
+ The additional advanced codec negotiation options
+ have also been removed from the sample configuration
+ and marked as reserved for future functionality in
+ XML documentation.
+
+ The codec preference options have also been fixed to
+ enforce local codec configuration.
+
+ ASTERISK-29109
+
+
+#### tcptls.c: Don't close TCP client file descriptors more than once
+ Author: Sean Bright
+ Date: 2020-09-30
+
+ ASTERISK-28430 #close
+
+
+#### resource_endpoints.c: memory leak when providing a 404 response
+ Author: Jean Aunis
+ Date: 2020-10-05
+
+ When handling a send_message request to a non-existing endpoint, the response's
+ body is overriden and not properly freed.
+
+ ASTERISK-29108
+
+
+#### Logging: Add debug logging categories
+ Author: Kevin Harwell
+ Date: 2020-08-28
+
+ Added debug logging categories that allow a user to output debug
+ information based on a specified category. This lets the user limit,
+ and filter debug output to data relevant to a particular context,
+ or topic. For instance the following categories are now available for
+ debug logging purposes:
+
+ dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet,
+ stun, stun_packet
+
+ These debug categories can be enable/disable via an Asterisk CLI command.
+
+ While this overrides, and outputs debug data, core system debugging is
+ not affected by this patch. Statements still output at their appropriate
+ debug level. As well backwards compatibility has been maintained with
+ past debug groups that could be enabled using the CLI (e.g. rtpdebug,
+ stundebug, etc.).
+
+ ASTERISK-29054 #close
+
+
+#### pbx.c: On error, ast_add_extension2_lockopt should always free 'data'
+ Author: Sean Bright
+ Date: 2020-09-29
+
+ In the event that the desired extension already exists,
+ ast_add_extension2_lockopt() will free the 'data' it is passed before
+ returning an error, so we should not be freeing it ourselves.
+
+ Additionally, there were two places where ast_add_extension2_lockopt()
+ could return an error without also freeing the 'data' pointer, so we
+ add that.
+
+ ASTERISK-29097 #close
+
+
+#### app_confbridge/bridge_softmix: Add ability to force estimated bitrate
+ Author: George Joseph
+ Date: 2020-09-24
+
+ app_confbridge now has the ability to set the estimated bitrate on an
+ SFU bridge. To use it, set a bridge profile's remb_behavior to "force"
+ and set remb_estimated_bitrate to a rate in bits per second. The
+ remb_estimated_bitrate parameter is ignored if remb_behavior is something
+ other than "force".
+
+
+#### app_voicemail.c: Document VMSayName interruption behavior
+ Author: Sean Bright
+ Date: 2020-09-29
+
+ ASTERISK-26424 #close
+
+
+#### res_pjsip_sdp_rtp: Fix accidentally native bridging calls
+ Author: Holger Hans Peter Freyther
+ Date: 2020-09-23
+
+ Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
+ auto but no tel_event was found inside SDP file.
+
+ On an incoming call create_rtp will be called and when session->dtmf is
+ set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
+ looking at the SDP file.
+
+ Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
+ but continued to advertise RFC2833 support.
+
+ This meant the native_rtp bridge would falsely consider the two channels
+ as compatible. In addition to changing the DTMF mode we now set or
+ remove the AST_RTP_PROPERTY_DTMF.
+
+ The property is checked in ast_rtp_dtmf_compatible and called by
+ native_rtp_bridge_compatible.
+
+ ASTERISK-29051 #close
+
+
+#### res_musiconhold: Load all realtime entries, not just the first
+ Author: lvl
+ Date: 2020-09-28
+
+ ASTERISK-29099
+
+
+#### channels: Don't dereference NULL pointer
+ Author: Jasper van der Neut
+ Date: 2020-09-23
+
+ Check result of ast_translator_build_path against NULL before dereferencing.
+
+ ASTERISK-29091
+
+
+#### res_pjsip_diversion: fix double 181
+ Author: Torrey Searle
+ Date: 2020-09-24
+
+ Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and
+ AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice,
+ resulting in to 181 being generated.
+
+
+#### res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs
+ Author: Sean Bright
+ Date: 2020-09-24
+
+
+#### dsp.c: Update calls to ast_format_cmp to check result properly
+ Author: Sean Bright
+ Date: 2020-09-23
+
+ ASTERISK-28311 #close
+
+
+#### res_pjsip_session: Fix stream name memory leak.
+ Author: Joshua C. Colp
+ Date: 2020-09-22
+
+ When constructing a stream name based on the media type
+ and position the allocated name was not being freed
+ causing a leak.
+
+
+#### func_curl.c: Prevent crash when using CURLOPT(httpheader)
+ Author: Sean Bright
+ Date: 2020-09-18
+
+ Because we use shared thread-local cURL instances, we need to ensure
+ that the state of the cURL instance is correct before each invocation.
+
+ In the case of custom headers, we were not resetting cURL's internal
+ HTTP header pointer which could result in a crash if subsequent
+ requests do not configure custom headers.
+
+ ASTERISK-29085 #close
+
+
+#### res_musiconhold: Start playlist after initial announcement
+ Author: Sean Bright
+ Date: 2020-09-18
+
+ Only track our sample offset if we are playing a non-announcement file,
+ otherwise we will skip that number of samples when we start playing the
+ first MoH file.
+
+ ASTERISK-24329 #close
+
+
+#### res_pjsip_session: Fix session reference leak.
+ Author: Joshua C. Colp
+ Date: 2020-09-22
+
+ The ast_sip_dialog_get_session function returns the session
+ with reference count increased. This was not taken into
+ account and was causing sessions to remain around when they
+ should not be.
+
+ ASTERISK-29089
+
+
+#### res_stasis.c: Add compare function for bridges moh container
+ Author: Michal Hajek
+ Date: 2020-09-16
+
+ Sometimes not play MOH on bridge.
+
+ ASTERISK-29081
+ Reported-by: Michal Hajek <michal.hajek@daktela.com>
+
+
+#### logger.h: Fix ast_trace to respect scope_level
+ Author: George Joseph
+ Date: 2020-09-17
+
+ ast_trace() was always emitting messages when it's level was set to -1
+ because it was ignoring scope_level.
+
+
+#### chan_sip.c: Don't build by default
+ Author: Sean Bright
+ Date: 2020-09-15
+
+ ASTERISK-29083 #close
+
+
+#### audiosocket: Fix module menuselect descriptions
+ Author: Sean Bright
+ Date: 2020-09-15
+
+ The module description needs to be on the same line as the
+ AST_MODULE_INFO or it is not parsed correctly.
+
+
+#### bridge_softmix/sfu_topologies_on_join: Ignore topology change failures
+ Author: George Joseph
+ Date: 2020-09-17
+
+ When a channel joins a bridge, we do topology change requests on all
+ existing channels to add the new participant to them. However the
+ announcer channel will return an error because it doesn't support
+ topology in the first place. Unfortunately, there doesn't seem to be a
+ reliable way to tell if the error is expected or not so the error is
+ ignored for all channels. If the request fails on a "real" channel,
+ that channel just won't get the new participant's video.
+
+
+#### res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined
+ Author: Sean Bright
+ Date: 2020-09-15
+
+
+#### res_pjsip_diversion: implement support for History-Info
+ Author: Torrey Searle
+ Date: 2020-08-13
+
+ Implemention of History-Info capable of interworking with Diversion
+ Header following RFC7544
+
+ ASTERISK-29027 #close
+
+
+#### format_cap: Perform codec lookups by pointer instead of name
+ Author: Sean Bright
+ Date: 2020-09-14
+
+ ASTERISK-28416 #close
+
+
+#### res_pjsip_session: Fix issue with COLP and 491
+ Author: George Joseph
+ Date: 2020-09-11
+
+ The recent 491 changes introduced a check to determine if the active
+ and pending topologies were equal and to suppress the re-invite if they
+ were. When a re-invite is sent for a COLP-only change, the pending
+ topology is NULL so that check doesn't happen and the re-invite is
+ correctly sent. Of course, sending the re-invite sets the pending
+ topology. If a 491 is received, when we resend the re-invite, the
+ pending topology is set and since we didn't request a change to the
+ topology in the first place, pending and active topologies are equal so
+ the topologies-equal check causes the re-invite to be erroneously
+ suppressed.
+
+ This change checks if the topologies are equal before we run the media
+ state resolver (which recreates the pending topology) so that when we
+ do the final topologies-equal check we know if this was a topology
+ change request. If it wasn't a change request, we don't suppress
+ the re-invite even though the topologies are equal.
+
+ ASTERISK-29014
+
+
+#### debugging: Add enough to choke a mule
+ Author: George Joseph
+ Date: 2020-08-20
+
+ Added to:
+ * bridges/bridge_softmix.c
+ * channels/chan_pjsip.c
+ * include/asterisk/res_pjsip_session.h
+ * main/channel.c
+ * res/res_pjsip_session.c
+
+ There NO functional changes in this commit.
+
+
+#### res_pjsip_session: Handle multi-stream re-invites better
+ Author: George Joseph
+ Date: 2020-08-20
+
+ When both Asterisk and a UA send re-invites at the same time, both
+ send 491 "Transaction in progress" responses to each other and back
+ off a specified amount of time before retrying. When Asterisk
+ prepares to send its re-invite, it sets up the session's pending
+ media state with the new topology it wants, then sends the
+ re-invite. Unfortunately, when it received the re-invite from the
+ UA, it partially processed the media in the re-invite and reset
+ the pending media state before sending the 491 losing the state it
+ set in its own re-invite.
+
+ Asterisk also was not tracking re-invites received while an existing
+ re-invite was queued resulting in sending stale SDP with missing
+ or duplicated streams, or no re-invite at all because we erroneously
+ determined that a re-invite wasn't needed.
+
+ There was also an issue in bridge_softmix where we were using a stream
+ from the wrong topology to determine if a stream was added. This also
+ caused us to erroneously determine that a re-invite wasn't needed.
+
+ Regardless of how the delayed re-invite was triggered, we need to
+ reconcile the topology that was active at the time the delayed
+ request was queued, the pending topology of the queued request,
+ and the topology currently active on the session. To do this we
+ need a topology resolver AND we need to make stream named unique
+ so we can accurately tell what a stream has been added or removed
+ and if we can re-use a slot in the topology.
+
+ Summary of changes:
+
+ * bridge_softmix:
+ * We no longer reset the stream name to "removed" in
+ remove_all_original_streams(). That was causing multiple streams
+ to have the same name and wrecked the checks for duplicate streams.
+
+ * softmix_bridge_stream_sources_update() was checking the old_stream
+ to see if it had the softmix prefix and not considering the stream
+ as "new" if it did. If the stream in that slot has something in it
+ because another re-invite happened, then that slot in old might
+ have a softmix stream but the same stream in new might actually
+ be a new one. Now we check the new_stream's name instead of
+ the old_stream's.
+
+ * stream:
+ * Instead of using plain media type name ("audio", "video", etc) as
+ the default stream name, we now append the stream position to it
+ to make it unique. We need to do this so we can distinguish multiple
+ streams of the same type from each other.
+
+ * When we set a stream's state to REMOVED, we no longer reset its
+ name to "removed" or destroy its metadata. Again, we need to
+ do this so we can distinguish multiple streams of the same
+ type from each other.
+
+ * res_pjsip_session:
+ * Added resolve_refresh_media_states() that takes in 3 media states
+ and creates an up-to-date pending media state that includes the changes
+ that might have happened while a delayed session refresh was in the
+ delayed queue.
+
+ * Added is_media_state_valid() that checks the consistency of
+ a media state and returns a true/false value. A valid state has:
+ * The same number of stream entries as media session entries.
+ Some media session entries can be NULL however.
+ * No duplicate streams.
+ * A valid stream for each non-NULL media session.
+ * A stream that matches each media session's stream_num
+ and media type.
+
+ * Updated handle_incoming_sdp() to set the stream name to include the
+ stream position number in the name to make it unique.
+
+ * Updated the ast_sip_session_delayed_request structure to include both
+ the pending and active media states and updated the associated delay
+ functions to process them.
+
+ * Updated sip_session_refresh() to accept both the pending and active
+ media states that were in effect when the request was originally queued
+ and to pass them on should the request need to be delayed again.
+
+ * Updated sip_session_refresh() to call resolve_refresh_media_states()
+ and substitute its results for the pending state passed in.
+
+ * Updated sip_session_refresh() with additional debugging.
+
+ * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE
+ to pjproject if a transaction is in progress. This stops us from
+ creating a partial pending media state that would be invalid later on.
+
+ * Updated reschedule_reinvite() to clone both the current pending and
+ active media states and pass them to delay_request() so the resolver
+ can tell what the original intention of the re-invite was.
+
+ * Added a large unit test for the resolver.
+
+ ASTERISK-29014
+
+
+#### realtime: Increased reg_server character size
+ Author: Sungtae Kim
+ Date: 2020-08-31
+
+ Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20).
+ This is fine for normal cases, but if the hostname is longer than 20, it returns error and then
+ failed to register the contact address of the peer.
+
+ Normally, 20 characters limitation for the hostname is fine, but with the cloud env.
+ So, increased the size to 255.
+
+ ASTERISK-29056
+
+
+#### res_stasis.c: Added video_single option for bridge creation
+ Author: Sungtae Kim
+ Date: 2020-08-30
+
+ Currently, it was not possible to create bridge with video_mode single.
+ This made hard to put the bridge in a vidoe_single mode.
+ So, added video_single option for Bridge creation using the ARI.
+ This allows create a bridge with video_mode single.
+
+ ASTERISK-29055
+
+
+#### Bridging: Use a ref to bridge_channel's channel to prevent crash.
+ Author: Ben Ford
+ Date: 2020-08-31
+
+ There's a race condition with bridging where a bridge can be torn down
+ causing the bridge_channel's ast_channel to become NULL when it's still
+ needed. This particular case happened with attended transfers, but the
+ crash occurred when trying to publish a stasis message. Now, the
+ bridge_channel is locked, a ref to the ast_channel is obtained, and that
+ ref is passed down the chain.
+
+
+#### res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a..
+ Author: Patrick Verzele
+ Date: 2020-09-01
+
+ Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again.
+
+
+#### conversions: Add string to signed integer conversion functions
+ Author: Kevin Harwell
+ Date: 2020-08-28
+
+
+#### app_queue: Fix leave-empty not recording a call as abandoned
+ Author: Kfir Itzhak
+ Date: 2020-08-26
+
+ This fixes a bug introduced mistakenly in ASTERISK-25665:
+ If leave-empty is enabled, a call may sometimes be removed from
+ a queue without recording it as abandoned.
+ This causes Asterisk to not generate an abandon event for that
+ call, and for the queue abandoned counter to be incorrect.
+
+ ASTERISK-29043 #close
+
+
+#### ast_coredumper: Fix issues with naming
+ Author: George Joseph
+ Date: 2020-08-28
+
+ If you run ast_coredumper --tarball-coredumps in the same directory
+ as the actual coredump, tar can fail because the link to the
+ actual coredump becomes recursive. The resulting tarball will
+ have everything _except_ the coredump (which is usually what
+ you need)
+
+ There's also an issue that the directory name in the tarball
+ is the same as the coredump so if you extract the tarball the
+ directory it creates will overwrite the coredump.
+
+ So:
+
+ * Made the link to the coredump use the absolute path to the
+ file instead of a relative one. This prevents the recursive
+ link and allows tar to add the coredump.
+
+ * The tarballed directory is now named <coredump>.output instead
+ of just <coredump> so if you expand the tarball it won't
+ overwrite the coredump.
+
+
+#### parking: Copy parker UUID as well.
+ Author: Joshua C. Colp
+ Date: 2020-08-28
+
+ When fixing issues uncovered by GCC10 a copy of the parker UUID
+ was removed accidentally. This change restores it so that the
+ subscription has the data it needs.
+
+ ASTERISK-29042
+
+
+#### sip_nat_settings: Update script for latest Linux.
+ Author: Alexander Traud
+ Date: 2020-08-26
+
+ With the latest Linux, 'ifconfig' is not installed on default anymore.
+ Furthermore, the output of the current net-tools 'ifconfig' changed.
+ Therefore, parsing failed. This update uses 'ip addr show' instead.
+ Finally, the service for the external IP changed.
+
+
+#### samples: Fix keep_alive_interval default in pjsip.conf.
+ Author: Alexander Traud
+ Date: 2020-08-26
+
+ Since ASTERISK_27978 the default is not off but 90 seconds. That change
+ happened because ASTERISK_27347 disabled the keep-alives in the bundled
+ PJProject and Asterisk should behave the same as before.
+
+
+#### chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution
+ Author: Kevin Harwell
+ Date: 2020-08-24
+
+ This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function
+ is called on a channel prior to answering a warning is issued and the
+ function returns unsuccessful.
+
+ ASTERISK-28878 #close
+
+
+#### pbx: Fix hints deadlock between reload and ExtensionState.
+ Author: Joshua C. Colp
+ Date: 2020-08-27
+
+ When the ExtensionState AMI action is executed on a pattern matched
+ hint it can end up adding a new hint if one does not already exist.
+ This results in a locking order of contexts -> hints -> contexts.
+
+ If at the same time a reload is occurring and adding its own hint
+ it will have a locking order of hints -> contexts.
+
+ This results in a deadlock as one thread wants a lock on contexts
+ that the other has, and the other thread wants a lock on hints
+ that the other has.
+
+ This change enforces a hints -> contexts locking order by explicitly
+ locking hints in the places where a hint is added when queried for.
+ This matches the order seen through normal adding of hints.
+
+ ASTERISK-29046
+
+
+#### logger.c: Added a new log formatter called "plain"
+ Author: George Joseph
+ Date: 2020-08-14
+
+ Added a new log formatter called "plain" that always prints
+ file, function and line number if available (even for verbose
+ messages) and never prints color control characters. It also
+ doesn't apply any special formatting for verbose messages.
+ Most suitable for file output but can be used for other channels
+ as well.
+
+ You use it in logger.conf like so:
+ debug => [plain]debug
+ console => [plain]error,warning,debug,notice,pjsip_history
+ messages => [plain]warning,error,verbose
+
+
+#### res_speech: Bump reference on format object
+ Author: Nickolay Shmyrev
+ Date: 2020-08-21
+
+ Properly bump reference on format object to avoid memory corruption on double free
+
+ ASTERISK-29040 #close
+
+
+#### res_pjsip_diversion: handle 181
+ Author: Torrey Searle
+ Date: 2020-07-22
+
+ Adapt the response handler so it also called when 181 is received.
+ In the case 181 is received, also generate the 181 response.
+
+ ASTERISK-29001 #close
+
+
+#### app_voicemail: Process urgent messages with mailcmd
+ Author: Sean Bright
+ Date: 2020-08-21
+
+ Rather than putting messages into INBOX and then moving them to Urgent
+ later, put them directly in to the Urgent folder. This prevents
+ mailcmd from being skipped.
+
+ ASTERISK-27273 #close
+
+
+#### app_queue: Member lastpause time reseting
+ Author: Evandro César Arruda
+ Date: 2020-08-21
+
+ This fixes the reseting members lastpause problem when realtime members is being used,
+ the function rt_handle_member_record was forcing the reset members lastpause because it
+ does not exist in realtime
+
+ ASTERISK-29034 #close
+
+
+#### res_pjsip_session: Don't aggressively terminate on failed re-INVITE.
+ Author: Joshua C. Colp
+ Date: 2020-08-18
+
+ Per the RFC when an outgoing re-INVITE is done we should
+ only terminate the dialog if a 481 or 408 is received.
+
+ ASTERISK-29033
+
+
+#### bridge_channel: Ensure text messages are zero terminated
+ Author: Sean Bright
+ Date: 2020-08-19
+
+ T.140 data in RTP is not zero terminated, so when we are queuing a text
+ frame on a bridge we need to ensure that we are passing a zero
+ terminated string.
+
+ ASTERISK-28974 #close
+
+
+#### res_musiconhold.c: Use ast_file_read_dir to scan MoH directory
+ Author: Sean Bright
+ Date: 2020-08-07
+
+ Two changes of note in this patch:
+
+ * Use ast_file_read_dir instead of opendir/readdir/closedir
+
+ * If the files list should be sorted, do that at the end rather than as
+ we go which improves performance for large lists
+
+
+#### scope_trace: Added debug messages and added additional macros
+ Author: George Joseph
+ Date: 2020-08-19
+
+ The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages
+ at the same level as the scope level. This allows the same
+ messages to be printed to the debug log when AST_DEVMODE
+ isn't enabled.
+
+ Also added a few variants of the SCOPE_EXIT macros that will
+ also call ast_log instead of ast_debug to make it easier to
+ use scope tracing and still print error messages.
+
+
+#### stream.c: Added 2 more debugging utils and added pos to stream string
+ Author: George Joseph
+ Date: 2020-08-20
+
+ * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros
+ which are shortcuts for
+ ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP))
+
+ * Added the stream position to the string representation of the
+ stream.
+
+ * Fixed some formatting in ast_stream_to_str().
+
+
+#### chan_sip: Clear ToHost property on peer when changing to dynamic host
+ Author: Dennis Buteyn
+ Date: 2020-02-18
+
+ The ToHost parameter was not cleared when a peer's host value was
+ changed to dynamic. This causes invites to be sent to the original host.
+
+ ASTERISK-29011 #close
+
+
+#### ACN: Changes specific to the core
+ Author: George Joseph
+ Date: 2020-07-20
+
+ Allow passing a topology from the called channel back to the
+ calling channel.
+
+ * Added a new function ast_queue_answer() that accepts a stream
+ topology and queues an ANSWER CONTROL frame with it as the
+ data. This allows the called channel to indicate its resolved
+ topology.
+
+ * Added a new virtual function to the channel tech structure
+ answer_with_stream_topology() that allows the calling channel
+ to receive the called channel's topology. Added
+ ast_raw_answer_with_stream_topology() that invokes that virtual
+ function.
+
+ * Modified app_dial.c and features.c to grab the topology from the
+ ANSWER frame queued by the answering channel and send it to
+ the calling channel with ast_raw_answer_with_stream_topology().
+
+ * Modified frame.c to automatically cleanup the reference
+ to the topology on ANSWER frames.
+
+ Added a few debugging messages to stream.c.
+
+
+#### Makefile: Fix certified version numbers
+ Author: cmaj
+ Date: 2020-08-06
+
+ Adds sed before awk to produce reasonable ASTERISKVERSIONNUM
+ on certified versions of Asterisk eg. 16.8-cert3 is 160803
+ instead of the previous 00800.
+
+ ASTERISK-29021 #close
+
+
+#### res_musiconhold.c: Prevent crash with realtime MoH
+ Author: Sean Bright
+ Date: 2020-08-06
+
+ The MoH class internal file vector is potentially being manipulated by
+ multiple threads at the same time without sufficient locking. Switch to
+ a reference counted list and operate on copies where necessary.
+
+ ASTERISK-28927 #close
+
+
+#### res_pjsip: Fix codec preference defaults.
+ Author: Joshua C. Colp
+ Date: 2020-08-06
+
+ When reading in a codec preference configuration option
+ the value would be set on the respective option before
+ applying any default adjustments, resulting in the
+ configuration not being as expected.
+
+ This was exposed by the REST API push configuration as
+ it used the configuration returned by Asterisk to then do
+ a modification. In the case of codec preferences one of
+ the options had a transcode value of "unspecified" when the
+ defaults should have ensured it would be "allow" instead.
+
+ This also renames the options in other places that were
+ missed.
+
+
+#### vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors
+ Author: Sean Bright
+ Date: 2020-08-04
+
+ The assumed behavior of realloc() - that it was effectively a free() if
+ its second argument was 0 - is Linux specific behavior and is not
+ guaranteed by either POSIX or the C specification.
+
+ Instead, if we want to resize a vector to 0, do it explicitly.
+
+
+#### pjproject: clone sdp to protect against (nat) modifications
+ Author: Michael Neuhauser
+ Date: 2020-06-30
+
+ PJSIP, UDP transport with external_media_address and session timers
+ enabled. Connected to SIP server that is not in local net. Asterisk
+ initiated the connection and is refreshing the session after 150s
+ (timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has
+ a malformed IP address in its SDP (garbage string). This only happens
+ when the SDP is modified by the nat-code to replace the local IP address
+ with the configured external_media_address.
+ Analysis: the code to modify the SDP (in
+ res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?)
+ in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses
+ the tdata->pool to allocate the replacement string. But the same
+ pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also
+ used for the 2nd refresh-INVITE (because it is stored in pjmedia's
+ pjmedia_sdp_neg structure). The problem is, that at that moment, the
+ tdata->pool that holds the stringified external_media_address from the
+ 1. refresh-INVITE has long been reused for something else.
+ Fix by Sauw Ming of pjproject (see
+ https://github.com/pjsip/pjproject/pull/2476): the local, potentially
+ modified pjmedia_sdp_stream is cloned in
+ pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the
+ clone is stored, thereby detaching from the tdata->pool (which is only
+ released *after* process_answer())
+
+ ASTERISK-28973
+ Reported-by: Michael Neuhauser
+
+
+#### utils.c: NULL terminate ast_base64decode_string.
+ Author: Ben Ford
+ Date: 2020-08-04
+
+ With the addition of STIR/SHAKEN, the function ast_base64decode_string
+ was added for convenience since there is a lot of converting done during
+ the STIR/SHAKEN process. This function returned the decoded string for
+ you, but did not NULL terminate it, causing some issues (specifically
+ with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the
+ documentation has been updated to reflect this.
+
+
+#### ACN: Configuration renaming for pjsip endpoint
+ Author: George Joseph
+ Date: 2020-07-21
+
+ This change renames the codec preference endpoint options.
+ incoming_offer_codec_prefs becomes codec_prefs_incoming_offer
+ to keep the options together when showing an endpoint.
+
+
+#### res_stir_shaken: Fix memory allocation error in curl.c
+ Author: Ben Ford
+ Date: 2020-07-20
+
+ Fixed a memory allocation that was not passing in the correct size for
+ the struct in curl.c.
+
+
+#### res_pjsip_session: Ensure reused streams have correct bundle group
+ Author: George Joseph
+ Date: 2020-07-23
+
+ When a bundled stream is removed, its bundle_group is reset to -1.
+ If that stream is later reused, the bundle parameters on session
+ media need to be reset correctly it could mistakenly be rebundled
+ with a stream that was removed and never reused. Since the removed
+ stream has no rtp instance, a crash will result.
+
+
+#### res_pjsip_registrar: Don't specify an expiration for static contacts.
+ Author: Joshua C. Colp
+ Date: 2020-07-22
+
+ Statically configured contacts on an AOR don't have an expiration
+ time so when adding them to the resulting 200 OK if an endpoint
+ registers ensure they are marked as such.
+
+ ASTERISK-28995
+
+
+#### utf8.c: Add UTF-8 validation and utility functions
+ Author: Sean Bright
+ Date: 2020-07-13
+
+ There are various places in Asterisk - specifically in regards to
+ database integration - where having some kind of UTF-8 validation would
+ be beneficial. This patch adds:
+
+ * Functions to validate that a given string contains only valid UTF-8
+ sequences.
+
+ * A function to copy a string (similar to ast_copy_string) stopping when
+ an invalid UTF-8 sequence is encountered.
+
+ * A UTF-8 validator that allows for progressive validation.
+
+ All of this is based on the excellent UTF-8 decoder by Björn Höhrmann.
+ More information is available here:
+
+ https://bjoern.hoehrmann.de/utf-8/decoder/dfa/
+
+ The API was written in such a way that should allow us to replace the
+ implementation later should we determine that we need something more
+ comprehensive.
+
+
+#### stasis_bridge.c: Fixed wrong video_mode shown
+ Author: sungtae kim
+ Date: 2020-07-11
+
+ Currently, if the bridge has created by the ARI, the video_mode
+ parameter was
+ not shown in the BridgeCreated event correctly.
+
+ Fixed it and added video_mode shown in the 'bridge show <bridge id>'
+ cli.
+
+ ASTERISK-28987
+
+
+#### vector.h: Add AST_VECTOR_SORT()
+ Author: Sean Bright
+ Date: 2020-07-20
+
+ Allows a vector to be sorted in-place, rather than only during
+ insertion.
+
+
+#### CI: Force publishAsteriskDocs to use python2
+ Author: George Joseph
+ Date: 2020-07-16
+
+
+#### websocket / pjsip: Increase maximum packet size.
+ Author: Joshua C. Colp
+ Date: 2020-07-22
+
+ When dealing with a lot of video streams on WebRTC
+ the resulting SDPs can grow to be quite large. This
+ effectively doubles the maximum size to allow more
+ streams to exist.
+
+ The res_http_websocket module has also been changed
+ to use a buffer on the session for reading in packets
+ to ensure that the stack space usage is not excessive.
+
+
+#### Prepare master for the next Asterisk version
+ Author: George Joseph
+ Date: 2020-07-15
+
+ * Updated AMI version to 8.0.0
+ * Updated ARI version to 7.0.0
+ * Update make_ari_stubs.py to "Asterisk 19"
+
+
+#### acl.c: Coerce a NULL pointer into the empty string
+ Author: Sean Bright
+ Date: 2020-07-13
+
+ If an ACL is misconfigured in the realtime database (for instance, the
+ "rule" is blank) and Asterisk attempts to read the ACL, Asterisk will
+ crash.
+
+ ASTERISK-28978 #close
+
+
+#### pjsip: Include timer patch to prevent cancelling timer 0.
+ Author: Joshua C. Colp
+ Date: 2020-07-13
+
+ I noticed this while looking at another issue and brought
+ it up with Teluu. It was possible for an uninitialized timer
+ to be cancelled, resulting in the invalid timer id of 0
+ being placed into the timer heap causing issues.
+
+ This change is a backport from the pjproject repository
+ preventing this from happening.
+
+
diff --git a/contrib/realtime/mysql/mysql_queue_log.sql b/contrib/realtime/mysql/mysql_queue_log.sql
index 13dde964c29793ddd8452b0f1a1dcf21ac971ac9..924f8406feb3f437eb958bcd30dd3f8d47a8bf37 100644
--- a/contrib/realtime/mysql/mysql_queue_log.sql
+++ b/contrib/realtime/mysql/mysql_queue_log.sql
@@ -23,7 +23,7 @@ CREATE TABLE queue_log (
UNIQUE (id)
);
-INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
+INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58') RETURNING alembic_version.version_num;
COMMIT;
diff --git a/contrib/realtime/postgresql/postgresql_cdr.sql b/contrib/realtime/postgresql/postgresql_cdr.sql
index 79380171a8bff739f7be0a7f326d50f02a6ba20d..409b90ef73467bc42d9d0ad92cad9d5e3b808edc 100644
--- a/contrib/realtime/postgresql/postgresql_cdr.sql
+++ b/contrib/realtime/postgresql/postgresql_cdr.sql
@@ -31,7 +31,7 @@ CREATE TABLE cdr (
sequence INTEGER
);
-INSERT INTO alembic_version (version_num) VALUES ('210693f3123d');
+INSERT INTO alembic_version (version_num) VALUES ('210693f3123d') RETURNING alembic_version.version_num;
-- Running upgrade 210693f3123d -> 54cde9847798
diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql
index dcf823b4d15b2c21848308c7d5183915f6c7b183..b2d7597ec5672260f19ae6775b1756a1096d0051 100644
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
@@ -272,7 +272,7 @@ CREATE TABLE musiconhold (
PRIMARY KEY (name)
);
-INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c');
+INSERT INTO alembic_version (version_num) VALUES ('4da0c5f79a9c') RETURNING alembic_version.version_num;
-- Running upgrade 4da0c5f79a9c -> 43956d550a44
diff --git a/contrib/realtime/postgresql/postgresql_queue_log.sql b/contrib/realtime/postgresql/postgresql_queue_log.sql
index 13dde964c29793ddd8452b0f1a1dcf21ac971ac9..924f8406feb3f437eb958bcd30dd3f8d47a8bf37 100644
--- a/contrib/realtime/postgresql/postgresql_queue_log.sql
+++ b/contrib/realtime/postgresql/postgresql_queue_log.sql
@@ -23,7 +23,7 @@ CREATE TABLE queue_log (
UNIQUE (id)
);
-INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58');
+INSERT INTO alembic_version (version_num) VALUES ('4105ee839f58') RETURNING alembic_version.version_num;
COMMIT;
diff --git a/contrib/realtime/postgresql/postgresql_voicemail.sql b/contrib/realtime/postgresql/postgresql_voicemail.sql
index 274749ae60cc71f73e0552a5c91829a21e5033d7..13fa0721252a43f8a75bc08fdd02c3a4a04b2c85 100644
--- a/contrib/realtime/postgresql/postgresql_voicemail.sql
+++ b/contrib/realtime/postgresql/postgresql_voicemail.sql
@@ -27,7 +27,7 @@ ALTER TABLE voicemail_messages ADD CONSTRAINT voicemail_messages_dir_msgnum PRIM
CREATE INDEX voicemail_messages_dir ON voicemail_messages (dir);
-INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e');
+INSERT INTO alembic_version (version_num) VALUES ('a2e9769475e') RETURNING alembic_version.version_num;
-- Running upgrade a2e9769475e -> 39428242f7f5