diff --git a/.version b/.version
index 80645cbd62fd238c5532d07fee89c250bb66f4a6..9f583440f7d3eaf761937ba4e830179737aed12c 100644
--- a/.version
+++ b/.version
@@ -1 +1 @@
-21.0.2
+21.1.0-rc1
diff --git a/CHANGES.md b/CHANGES.md
index 0d1e807b63ff39725d7fef3fb4062c06d91aa7f7..0686d5929fce85a2a1c65fb24b8d7e04ccaec993 120000
--- a/CHANGES.md
+++ b/CHANGES.md
@@ -1 +1 @@
-ChangeLogs/ChangeLog-21.0.2.md
\ No newline at end of file
+ChangeLogs/ChangeLog-21.1.0-rc1.md
\ No newline at end of file
diff --git a/ChangeLogs/ChangeLog-21.1.0-rc1.md b/ChangeLogs/ChangeLog-21.1.0-rc1.md
new file mode 100644
index 0000000000000000000000000000000000000000..e785400bb1548abaf8821416817d1763d573f6dd
--- /dev/null
+++ b/ChangeLogs/ChangeLog-21.1.0-rc1.md
@@ -0,0 +1,1620 @@
+
+Change Log for Release asterisk-21.1.0-rc1
+========================================
+
+Links:
+----------------------------------------
+
+ - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0-rc1.md)
+ - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0-rc1)
+ - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0-rc1.tar.gz)
+ - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
+
+Summary:
+----------------------------------------
+
+- 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.
+- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
+- 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
+- pbx_config.c: Don't crash when unloading module.
+- 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_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
+- .github: PRSubmitActions: Fix adding reviewers to PR
+- .github: New PR Submit workflows
+- .github: New PR Submit workflows
+- 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.
+- .github: Fix job prereqs in PROpenedUpdated
+- .github: Block PR tests until approved
+- .github: Use generic releaser
+- 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
+- res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
+- 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
+- res_pjsip_pubsub: Add body_type to test_handler for unit tests
+- 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
+
+User Notes:
+----------------------------------------
+
+- ### 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 characters
+ 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.
+
+
+Upgrade Notes:
+----------------------------------------
+
+- ### 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 characters
+ 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.
+
+
+Closed Issues:
+----------------------------------------
+
+ - #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
+ - #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
+ - #242: [new-feature]: logger: Allow filtering logs in CLI by channel
+ - #248: [bug]: core_local: Local channels cannot have slashes in the destination
+ - #260: [bug]: maxptime must be changed to multiples of 20
+ - #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
+ - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree
+ - #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
+ - #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
+ - #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
+ - #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
+ - #482: [improvement]: manager.c: Improve clarity of "manager show connected" output
+ - #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.
+ - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels
+
+Commits By Author:
+----------------------------------------
+
+- ### Bastian Triller (1):
+ - func_json: Fix crashes for some types
+
+- ### Brad Smith (2):
+ - res_rtp_asterisk.c: Fix runtime issue with LibreSSL
+ - main/utils: Implement ast_get_tid() for OpenBSD
+
+- ### Eduardo (1):
+ - codec_builtin: Use multiples of 20 for maximum_ms
+
+- ### George Joseph (26):
+ - 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
+ - res_pjsip_pubsub: Add body_type to test_handler for unit tests
+ - 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
+ - .github: Use generic releaser
+ - .github: Block PR tests until approved
+ - .github: Fix job prereqs in PROpenedUpdated
+ - .github: New PR Submit workflows
+ - .github: New PR Submit workflows
+ - .github: PRSubmitActions: Fix adding reviewers to PR
+ - 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
+ - doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
+ - Revert "core & res_pjsip: Improve topology change handling."
+
+- ### Holger Hans Peter Freyther (3):
+ - 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
+
+- ### Jaco Kroon (1):
+ - app_queue: periodic announcement configurable start time.
+
+- ### Joshua C. Colp (1):
+ - variables: Add additional variable dialplan functions.
+
+- ### Mark Murawski (1):
+ - Remove files that are no longer updated
+
+- ### Matthew Fredrickson (2):
+ - app_followme.c: Grab reference on nativeformats before using it
+ - res_odbc.c: Allow concurrent access to request odbc connections
+
+- ### Maximilian Fridrich (3):
+ - 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
+
+- ### Mike Bradeen (7):
+ - 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
+
+- ### Naveen Albert (20):
+ - 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.
+
+- ### Samuel Olaechea (1):
+ - configs: Fix typo in pjsip.conf.sample.
+
+- ### Sean Bright (24):
+ - 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.
+ - pbx_config.c: Don't crash when unloading module.
+ - make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
+
+- ### Tinet-mucw (1):
+ - res_pjsip_transport_websocket: Prevent transport from being destroyed before message finishes.
+
+- ### Vitezslav Novy (1):
+ - res_rtp_asterisk: fix wrong counter management in ioqueue objects
+
+- ### sungtae kim (1):
+ - res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters
+
+
+Detail:
+----------------------------------------
+
+- ### 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
+
+- ### doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
+ Author: George Joseph
+ Date: 2023-12-15
+
+ These should have been deleted after the release of 21.0.0
+ but were missed.
+
+
+- ### 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.
+
+
+- ### pbx_config.c: Don't crash when unloading module.
+ Author: Sean Bright
+ Date: 2023-12-02
+
+ `pbx_config` subscribes to manager events to capture the `FullyBooted`
+ event but fails to unsubscribe if the module is loaded after that
+ event fires. If the module is unloaded, a crash occurs the next time a
+ manager event is raised.
+
+ We now unsubscribe when the module is unloaded if we haven't already
+ unsubscribed.
+
+ Fixes #470
+
+
+- ### 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 characters
+ 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.
+
+
+- ### .github: PRSubmitActions: Fix adding reviewers to PR
+ Author: George Joseph
+ Date: 2023-10-19
+
+
+- ### .github: New PR Submit workflows
+ Author: George Joseph
+ Date: 2023-10-17
+
+ The workflows that get triggered when PRs are submitted or updated
+ have been replaced with ones that are more secure and have
+ a higher level of parallelism.
+
+
+- ### .github: New PR Submit workflows
+ Author: George Joseph
+ Date: 2023-10-17
+
+ The workflows that get triggered when PRs are submitted or updated
+ have been replaced with ones that are more secure and have
+ a higher level of parallelism.
+
+
+- ### 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
+
+
+- ### .github: Fix job prereqs in PROpenedUpdated
+ Author: George Joseph
+ Date: 2023-10-09
+
+
+- ### .github: Block PR tests until approved
+ Author: George Joseph
+ Date: 2023-10-05
+
+
+- ### .github: Use generic releaser
+ Author: George Joseph
+ Date: 2023-08-15
+
+
+- ### 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 message finishes.
+ 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
+
+- ### res_pjsip_pubsub: Add body_type to test_handler for unit tests
+ Author: George Joseph
+ Date: 2023-09-15
+
+ The ast_sip_subscription_handler "test_handler" used for the unit
+ tests didn't set "body_type" so the NULL value was causing
+ a SEGV in build_subscription_tree(). It's now set to "".
+
+ Resolves: #335
+
+- ### 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
+
+
diff --git a/contrib/realtime/mysql/mysql_config.sql b/contrib/realtime/mysql/mysql_config.sql
index 8b71667e720df8d7dfa87e2c1b9dc289cc0cae6b..8628aa97af423d4d23aa71e31ae43380d88a9986 100644
--- a/contrib/realtime/mysql/mysql_config.sql
+++ b/contrib/realtime/mysql/mysql_config.sql
@@ -1418,3 +1418,59 @@ ALTER TABLE musiconhold ADD COLUMN loop_last ENUM('yes','no');
UPDATE alembic_version SET version_num='f5b0e7427449' WHERE alembic_version.version_num = '4042a0ff4d9f';
+-- Running upgrade f5b0e7427449 -> dac2b4c328b8
+
+ALTER TABLE ps_aors MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_aors MODIFY outbound_proxy VARCHAR(255) NULL;
+
+ALTER TABLE ps_auths MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_auths MODIFY realm VARCHAR(255) NULL;
+
+ALTER TABLE ps_contacts MODIFY outbound_proxy VARCHAR(255) NULL;
+
+ALTER TABLE ps_contacts MODIFY endpoint VARCHAR(255) NULL;
+
+ALTER TABLE ps_domain_aliases MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_domain_aliases MODIFY domain VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoint_id_ips MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoint_id_ips MODIFY endpoint VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoints MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoints MODIFY aors VARCHAR(2048) NULL;
+
+ALTER TABLE ps_endpoints MODIFY auth VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoints MODIFY outbound_auth VARCHAR(255) NULL;
+
+ALTER TABLE ps_endpoints MODIFY outbound_proxy VARCHAR(255) NULL;
+
+ALTER TABLE ps_inbound_publications MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_inbound_publications MODIFY endpoint VARCHAR(255) NULL;
+
+ALTER TABLE ps_outbound_publishes MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_outbound_publishes MODIFY outbound_auth VARCHAR(255) NULL;
+
+ALTER TABLE ps_registrations MODIFY id VARCHAR(255) NULL;
+
+ALTER TABLE ps_registrations MODIFY outbound_auth VARCHAR(255) NULL;
+
+ALTER TABLE ps_registrations MODIFY outbound_proxy VARCHAR(255) NULL;
+
+ALTER TABLE ps_registrations MODIFY endpoint VARCHAR(255) NULL;
+
+UPDATE alembic_version SET version_num='dac2b4c328b8' WHERE alembic_version.version_num = 'f5b0e7427449';
+
+-- Running upgrade dac2b4c328b8 -> 37a5332640e2
+
+ALTER TABLE ps_transports MODIFY method ENUM('default','unspecified','tlsv1','tlsv1_1','tlsv1_2','tlsv1_3','sslv2','sslv23','sslv3') NULL;
+
+UPDATE alembic_version SET version_num='37a5332640e2' WHERE alembic_version.version_num = 'dac2b4c328b8';
+
diff --git a/contrib/realtime/postgresql/postgresql_config.sql b/contrib/realtime/postgresql/postgresql_config.sql
index 47e1a76cfe07bb241764d0c0f6f56679cf562aa5..bc48664367652d65b4e10d07304f4e62e95a8a8b 100644
--- a/contrib/realtime/postgresql/postgresql_config.sql
+++ b/contrib/realtime/postgresql/postgresql_config.sql
@@ -1538,5 +1538,65 @@ ALTER TABLE musiconhold ADD COLUMN loop_last yesno_values;
UPDATE alembic_version SET version_num='f5b0e7427449' WHERE alembic_version.version_num = '4042a0ff4d9f';
+-- Running upgrade f5b0e7427449 -> dac2b4c328b8
+
+ALTER TABLE ps_aors ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_aors ALTER COLUMN outbound_proxy TYPE VARCHAR(255);
+
+ALTER TABLE ps_auths ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_auths ALTER COLUMN realm TYPE VARCHAR(255);
+
+ALTER TABLE ps_contacts ALTER COLUMN outbound_proxy TYPE VARCHAR(255);
+
+ALTER TABLE ps_contacts ALTER COLUMN endpoint TYPE VARCHAR(255);
+
+ALTER TABLE ps_domain_aliases ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_domain_aliases ALTER COLUMN domain TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoint_id_ips ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoint_id_ips ALTER COLUMN endpoint TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoints ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoints ALTER COLUMN aors TYPE VARCHAR(2048);
+
+ALTER TABLE ps_endpoints ALTER COLUMN auth TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoints ALTER COLUMN outbound_auth TYPE VARCHAR(255);
+
+ALTER TABLE ps_endpoints ALTER COLUMN outbound_proxy TYPE VARCHAR(255);
+
+ALTER TABLE ps_inbound_publications ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_inbound_publications ALTER COLUMN endpoint TYPE VARCHAR(255);
+
+ALTER TABLE ps_outbound_publishes ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_outbound_publishes ALTER COLUMN outbound_auth TYPE VARCHAR(255);
+
+ALTER TABLE ps_registrations ALTER COLUMN id TYPE VARCHAR(255);
+
+ALTER TABLE ps_registrations ALTER COLUMN outbound_auth TYPE VARCHAR(255);
+
+ALTER TABLE ps_registrations ALTER COLUMN outbound_proxy TYPE VARCHAR(255);
+
+ALTER TABLE ps_registrations ALTER COLUMN endpoint TYPE VARCHAR(255);
+
+UPDATE alembic_version SET version_num='dac2b4c328b8' WHERE alembic_version.version_num = 'f5b0e7427449';
+
+-- Running upgrade dac2b4c328b8 -> 37a5332640e2
+
+CREATE TYPE pjsip_transport_method_values_v2 AS ENUM ('default', 'unspecified', 'tlsv1', 'tlsv1_1', 'tlsv1_2', 'tlsv1_3', 'sslv2', 'sslv23', 'sslv3');
+
+ALTER TABLE ps_transports ALTER COLUMN method TYPE pjsip_transport_method_values_v2 USING method::text::pjsip_transport_method_values_v2;
+
+DROP TYPE pjsip_transport_method_values;
+
+UPDATE alembic_version SET version_num='37a5332640e2' WHERE alembic_version.version_num = 'dac2b4c328b8';
+
COMMIT;