# changelog

author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.2-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.5.2.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.5.2
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - res_rtp_asterisk: Fix regression issues with DTLS client check
  -
  - User Notes:
  - -----------
  -
  - Upgrade Notes:
  - --------------
  -
  - Closed Issues:
  - --------------
  -
  - #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()


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.6.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.6.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.6.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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_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
  - 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
  - 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
  - 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
  - #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
  - #539: [bug]: Existence of logger.xml causes linking failure


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.7.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.7.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.7.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.7.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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.
  -
  - User Notes:
  - -----------
  -
  - ### 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
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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.
  -
  - Closed Issues:
  - --------------
  -
  - #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
  - #351: [improvement]: Refactor res_stir_shaken to use libjwt
  - #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
  - #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
  - #492: [improvement]: res_calendar_icalendar: Print icalendar error if
  -       available on parsing failure
  - #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
  - #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
  -                      channel frame queue
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.8.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.8.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.8.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.8.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 44
  - Commit Authors: 15
  - Issues Resolved: 26
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip_logger: Preserve logging state on reloads.
  -                        Issuing "pjsip reload" will no longer disable logging
  -                        if it was previously enabled from the CLI.
  -
  - #### loader.c: Allow dependent modules to be unloaded recursively.
  -                In certain circumstances, modules with dependency relations can
  -                have their dependents automatically recursively unloaded and
  -                loaded again using the "module refresh" CLI command or the
  -                ModuleLoad AMI command.
  -
  - #### tcptls/iostream: Add support for setting SNI on client TLS connections
  -                       Secure websocket client connections now send SNI in the
  -                       TLS client hello.
  -
  - #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
  -                                        address.
  -                                        set identify_by=transport for the pjsip
  -                                        endpoint. Then use the existing 'match'
  -                                        option and the new 'transport' option
  -                                        of the identify. Fixes: #672
  -
  - #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  -                                        this new feature let users match
  -                                        endpoints based on the indound SIP
  -                                        requests' URI. To do so, add
  -                                        'request_uri' to the endpoint's
  -                                        'identify_by' option. The
  -                                        'match_request_uri' option of the
  -                                        identify can be an exact match for the
  -                                        entire request uri, or a regular
  -                                        expression (between slashes). It's
  -                                        quite similar to the header identifer.
  -                                        Fixes: #599
  -
  - #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  -                         the GET_TRANSFERRER_DATA dialplan variable can now be
  -                         used also in pjsip.
  -
  - #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  -                 When using the Originate AMI Action, we now can pass the
  -                 PreDialGoSub parameter, instructing the asterisk to perform an
  -                 subrouting at channel before call start. With this parameter
  -                 an call initiated by AMI can request the channel to start the
  -                 call automaticaly, adding a SIP header to using GoSUB,
  -                 instructing to autoanswer the channel, and proceeding the
  -                 outbuound extension executing. Exemple of an context to
  -                 perform the previus indication:
  -                 [addautoanswer]
  -                 exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  -                 exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  -                 exten => _s,n,Return()
  -
  - #### manager.c: Add CLI command to kick AMI sessions.
  -                 The "manager kick session" CLI command now allows kicking a
  -                 specified AMI session.
  -
  - #### chan_dahdi: Allow specifying waitfordialtone per call.
  -                  "waitfordialtone" may now be specified for DAHDI trunk
  -                  channels on a per-call basis using the CHANNEL function.
  -
  - #### Upgrade bundled pjproject to 2.14.1
  -      Bundled pjproject has been upgraded to 2.14.1. For more information visit
  -      pjproject Github page:
  -      https://github.com/pjsip/pjproject/releases/tag/2.14.1
  -
  - 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.
  -
  - Commit Authors:
  - ---------------
  -
  - Fabrice Fontaine: (1)
  - George Joseph: (8)
  - Henrik Liljedahl: (1)
  - Holger Hans Peter Freyther: (1)
  - Ivan Poddubny: (2)
  - Jonatascalebe: (1)
  - Joshua Elson: (1)
  - Martin Nystroem: (1)
  - Martin Tomec: (1)
  - Maximilian Fridrich: (1)
  - Naveen Albert: (14)
  - Sean Bright: (8)
  - Sperl Viktor: (2)
  - Spiridonov Dmitry: (1)
  - Stanislav Abramenkov: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.8.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.8.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.8.1
  -
  - Change Log for Release asterisk-20.8.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
  -    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
  -    requests
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.9.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.9.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.9.0
  - and https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.9.0
  -
  - This release resolves issues reported by the community and would have not been
  -  possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.9.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  - Commits: 20
  - Commit Authors: 9
  - Issues Resolved: 8
  - Security Advisories Resolved: 0
  -
  - 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.
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log
  -                 Add a Queue option log-restricted-caller-id to control whether
  -                 the Restricted Caller ID will be stored in the queue log. If
  -                 log-restricted-caller-id=no then the Caller ID will be stripped
  -                 if the Caller ID is restricted.
  -
  - #### pbx.c: expand fields width of "core show hints"
  -             The fields width of "core show hints" were increased. The width of
  -             "extension" field to 30 characters and the width of the "device
  -             state id" field to 60 characters.
  -
  - #### rtp_engine: add support for multirate RFC2833 digits
  -                  No change in configuration is required in order to enable this
  -                  feature. Endpoints configured to use RFC2833 will
  -                  automatically have this enabled. If the endpoint does not
  -                  support this, it should not include it in the SDP
  -                  offer/response.
  -                  Resolves: #699
  -
  - Upgrade Notes:
  - --------------
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log
  -                 Add a new column to the queues table:
  -                 queue_log_option_log_restricted
  -                 ENUM('0','1','off','on','false','true','no','yes') to control
  -                 whether the Restricted Caller ID will be stored in the queue
  -                 log.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - Bastian Triller: (1)
  - Chrsmj: (1)
  - George Joseph: (4)
  - Igor Goncharovsky: (1)
  - Mike Bradeen: (2)
  - Sean Bright: (7)
  - Tinet-Mucw: (1)
  - Walter Doekes: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.9.1-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.9.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.9.1
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.9.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 2
  - Commit Authors: 1
  - Issues Resolved: 2
  - Security Advisories Resolved: 0
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (2)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
  -             "make samples"
  - 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -> 20.9.0
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (2):
  -   - voicemail.conf.sample: Fix ':' comment typo
  -   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  -
  - Commit List:
  - ------------
  -
  -  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  -  voicemail.conf.sample: Fix ':' comment typo
  -
  - Commit Details:
  - ---------------
  -
  - #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  -                    Author: George Joseph
  -                    Date:   2024-07-25
  -
  -                    There can be empty slots in payload_mapping_tx corresponding
  -                    to dynamic payload types that haven't been seen before so we
  -                    now check for NULL before attempting to use 'type' in the
  -                    call to ast_format_cmp.
  -
  -                    Note: Currently only chan_sip calls
  -                    ast_rtp_codecs_payloads_unset()
  -
  -                    Resolves: #822
  -
  - #### voicemail.conf.sample: Fix ':' comment typo
  -                             Author: George Joseph
  -                             Date:   2024-07-24
  -
  -                             ...and removed an errant trailing space.
  -
  -                             Resolves: #819


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.9.2-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.9.2.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.9.2
  -
  - Change Log for Release asterisk-20.9.2
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  - - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
  -   Write=originate, is sufficient permissions for code execution / System()
  -   dialplan
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
  -                       execution / System() dialplan
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (1):
  -   - manager.c: Add entries to Originate blacklist
  -
  - Commit List:
  - ------------
  -
  - manager.c: Add entries to Originate blacklist
  -
  - Commit Details:
  - ---------------
  -
  - #### manager.c: Add entries to Originate blacklist
  -                 Author: George Joseph
  -                 Date:   2024-07-22
  -
  -                 Added Reload and DBdeltree to the list of dialplan application
  -                 that can't be executed via the Originate manager action without
  -                 also having write SYSTEM permissions.
  -
  -                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
  -                 dialplan functions that can't be executed via the Originate
  -                 manager action without also having write SYSTEM permissions.
  -
  -                 If the Queue application is attempted to be run by the
  -                 Originate manager action and an AGI parameter is specified in
  -                 the app data, it'll be rejected unless the manager user has
  -                 either the AGI or SYSTEM permissions.
  -
  -                 Resolves: #GHSA-c4cg-9275-6w44


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.9.3-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.9.3.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.9.3
  - and https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.9.3
  -
  - Change Log for Release asterisk-20.9.3
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
  -    A malformed Contact or Record-Route URI in an incoming SIP request can cause
  -    Asterisk to crash when res_resolver_unbound is used
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
  -                       SIP request can cause Asterisk to crash when
  -                       res_resolver_unbound is used
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (1):
  -   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -
  - Commit List:
  - ------------
  -
  - res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -
  - Commit Details:
  - ---------------
  -
  - #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -                            Author: George Joseph
  -                            Date:   2024-08-12
  -
  -                            The ub_result pointer passed to
  -                            unbound_resolver_callback by libunbound can be NULL
  -                            if the query was for something malformed like `.1`
  -                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
  -                            result and return instead of crashing with a SEGV.
  -                            This causes pjproject to simply cancel the
  -                            transaction with a "No answer record in the DNS
  -                            response" error. The existing "off nominal" unit
  -                            test was also updated to check this condition.
  -
  -                            Although not necessary for this fix, we also made
  -                            ast_dns_resolver_completed() tolerant of a NULL
  -                            result.
  -
  -                            Resolves: GHSA-v428-g3cw-7hv9


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.10.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.10.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.10.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.10.0
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 24
  - Commit Authors: 9
  - Issues Resolved: 18
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip_notify: add dialplan application
  -                        A new dialplan application PJSIPNotify is now available
  -                        which can send SIP NOTIFY requests from the dialplan.
  -                        The pjsip send notify CLI command has also been enhanced
  -                        to allow sending NOTIFY messages to a specific channel.
  -                        Syntax: pjsip send notify <option> channel <channel>
  -
  - #### channel: Add multi-tenant identifier.
  -               tenantid has been added to channels. It can be read in dialplan
  -               via CHANNEL(tenantid), and it can be set using
  -               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
  -               recommended to use the new tenantid option for pjsip endpoints
  -               (e.g., tenantid=My tenant ID) so that it will show up in
  -               Newchannel events. You can set it like any other channel variable
  -               using set_var in pjsip.conf as well, but note that this will NOT
  -               show up in Newchannel events. Tenant ID is also available in CDR
  -               and can be accessed with CDR(tenantid). The peer tenant ID can
  -               also be accessed with CDR(peertenantid). CEL includes tenant ID
  -               as well if it has been set.
  -
  - #### res_pjsip_config_wizard.c: Refactor load process
  -                                 The res_pjsip_config_wizard.so module can now
  -                                 be reloaded.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### channel: Add multi-tenant identifier.
  -               A new versioned struct (ast_channel_initializers) has been added
  -               that gets passed to __ast_channel_alloc_ap. The new function
  -               ast_channel_alloc_with_initializers should be used when creating
  -               channels that require the use of this struct. Currently the only
  -               value in the struct is for tenantid, but now more fields can be
  -               added to the struct as necessary rather than the
  -               __ast_channel_alloc_ap function. A new option (tenantid) has been
  -               added to endpoints in pjsip.conf as well. CEL has had its version
  -               bumped to include tenant ID.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - Ben Ford: (1)
  - Cade Parker: (1)
  - George Joseph: (10)
  - Jaco Kroon: (1)
  - Jean-Denis Girard: (1)
  - Mike Bradeen: (3)
  - Sean Bright: (2)
  - Tinet-Mucw: (3)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.11.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.11.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.11.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.11.0
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 40
  - Commit Authors: 9
  - Issues Resolved: 22
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
  -                 The new "suppress_moh_on_sendonly" endpoint option can be used
  -                 to prevent playing MOH back to a caller if the remote end sends
  -                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
  -
  - #### app_mixmonitor: Add 'D' option for dual-channel audio.
  -                      The MixMonitor application now has a new 'D' option which
  -                      interleaves the recorded audio in the output frames. This
  -                      allows for stereo recording output with one channel being
  -                      the transmitted audio and the other being the received
  -                      audio. The 't' and 't' options are compatible with this.
  -
  - #### manager.c: Restrict ModuleLoad to the configured modules directory.
  -                 The ModuleLoad AMI action now restricts modules to the
  -                 configured modules directory.
  -
  - #### manager: Enhance event filtering for performance
  -               You can now perform more granular filtering on events in
  -               manager.conf using expressions like
  -               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
  -               = PJSIP/` This is much more efficient than `eventfilter = Event:
  -               Newchannel.*Channel: PJSIP/` Full syntax guide is in
  -               configs/samples/manager.conf.sample.
  -
  - #### db.c: Remove limit on family/key length
  -            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
  -            removed and will now accept families and keys with a total length of
  -            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
  -            dialplan applications, dialplan functions, manager actions and
  -            `databse` CLI commands. Since the media_cache also uses the
  -            `ast_db_*()` APIs, you can now store resources with URIs longer than
  -            253 bytes.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (3)
  - Chrsmj: (1)
  - George Joseph: (15)
  - Jiangxc: (1)
  - Naveen Albert: (7)
  - Peter Jannesen: (2)
  - Sean Bright: (8)
  - Thomas Guebels: (2)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.11.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.11.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.11.1
  -
  - Change Log for Release asterisk-20.11.1
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
  -    Path traversal via AMI ListCategories allows access to outside files
  -
  - User Notes:
  - -----------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory.
  -                 The ListCategories AMI action now restricts files to the
  -                 configured configuration directory.
  -
  - Commit Authors:
  - ---------------
  -
  - Ben Ford: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
  -                       outside files
  -
  - Commits By Author:
  - ------------------
  -
  - #### Ben Ford (1):
  -  - manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit List:
  - ------------
  -
  -  manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit Details:
  - ---------------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory.
  -                 Author: Ben Ford
  -                 Date:   2024-12-17
  -
  -                 When using the ListCategories AMI action, it was possible to
  -                 traverse upwards through the directories to files outside of
  -                 the configured configuration directory. This action is now
  -                 restricted to the configured directory and an error will now be
  -                 returned if the specified file is outside of this limitation.
  -
  -  Resolves: #GHSA-33x6-fj46-6rfh
  -
  - UserNote: The ListCategories AMI action now restricts files to the configured
  -           configuration directory.


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.12.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.12.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.12.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.12.0
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 53
  - Commit Authors: 20
  - Issues Resolved: 19
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### sig_analog: Add Last Number Redial feature.
  -                  Users can now redial the last number called if the
  -                  lastnumredial setting is set to yes.
  -                  Resolves: #437
  -
  - #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
  -      The SHA-256 and SHA-512-256 algorithms are now available for
  -      authentication as both a UAS and a UAC.
  -
  - #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
  -      Bundled pjproject has been upgraded to 2.15.1. For more information visit
  -      pjproject Github page:
  -      https://github.com/pjsip/pjproject/releases/tag/2.15.1
  -
  - #### res_pjsip: Add new AOR option "qualify_2xx_only"
  -                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
  -                 that can be set so that only 2XX responses to OPTIONS requests
  -                 used to qualify a contact will mark the contact as available.
  -
  - #### app_queue: allow dynamically adding a queue member in paused state.
  -                 Use the p option of AddQueueMember() for paused member state.
  -                 Optionally, use the r(reason) option to specify a custom reason
  -                 for the pause.
  -
  - #### manager.c: Add Processed Call Count to CoreStatus output
  -                 The current processed call count is now returned as
  -                 CoreProcessedCalls from the CoreStatus AMI Action.
  -
  - #### func_curl.c: Add additional CURL options for SSL requests
  -                   The following new configuration options are now available in
  -                   the res_curl.conf file, and the CURL() function:
  -                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
  -                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
  -                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
  -                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
  -                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
  -                   See the libcurl documentation for more details.
  -
  - #### res_stir_shaken: Allow sending Identity headers for unknown TNs
  -                       You can now set the "unknown_tn_attest_level" option in
  -                       the attestation and/or profile objects in
  -                       stir_shaken.conf to enable sending Identity headers for
  -                       callerid TNs not explicitly configured.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### alembic: Database updates required.
  -               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
  -               authentication digest algorithms' 'res_pjsip: Add new AOR option
  -               "qualify_2xx_only"' ...have modified alembic scripts for the
  -               following database tables: ps_aors, ps_contacts, ps_auths,
  -               ps_globals. If you don't use the scripts to update your database,
  -               reads from those tables will succeeed but inserts into the
  -               ps_contacts table by res_pjsip_registrar will fail.
  -
  - Commit Authors:
  - ---------------
  -
  - Abdelkader Boudih: (3)
  - Alexey Khabulyak: (1)
  - Alexey Vasilyev: (1)
  - Allan Nathanson: (2)
  - Artem Umerov: (1)
  - George Joseph: (17)
  - Jaco Kroon: (1)
  - James Terhune: (1)
  - Joshua C. Colp: (1)
  - Kent: (1)
  - Maksim Nesterov: (1)
  - Maximilian Fridrich: (1)
  - Mike Pultz: (3)
  - Naveen Albert: (6)
  - Sean Bright: (6)
  - Sperl Viktor: (2)
  - Stanislav Abramenkov: (2)
  - Steffen Arntz: (1)
  - Tinet-Mucw: (1)
  - Viktor Litvinov: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.13.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.13.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.13.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.13.0
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 29
  - Commit Authors: 12
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### ari/pjsip: Make it possible to control transfers through ARI
  -                 Call transfers on the PJSIP channel can now be controlled by
  -                 ARI. This can be enabled by using the
  -                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (1)
  - Fabriziopicconi: (1)
  - George Joseph: (11)
  - Holger Hans Peter Freyther: (1)
  - Jeremy Lainé: (1)
  - Joshua Elson: (1)
  - Luz Paz: (3)
  - Maximilian Fridrich: (1)
  - Mike Bradeen: (1)
  - Naveen Albert: (1)
  - Sean Bright: (6)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.14.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.14.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 20.14.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.14.0
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 24
  - Commit Authors: 18
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
  -                   A Dial timeout on POST /channels/{channelId}/dial will now
  -                   result in a CANCEL and ChannelDestroyed with cause 19 / User
  -                   alerting, no answer. Previously no explicit cause was set,
  -                   resulting in a cause of 16 / Normal Call Clearing.
  -
  - contrib: Add systemd service and timer files for malloc trim.
  -          Service and timer files for systemd have been added to the
  -          contrib/systemd/ directory. If you are experiencing memory issues,
  -          install these files to have "malloc trim" periodically run on the
  -          system.
  -
  - Add log-caller-id-name option to log Caller ID Name in queue log
  -     This patch adds a global configuration option, log-caller-id-name, to
  -     queues.conf to control whether the Caller ID name is logged as parameter 4
  -     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
  -     is included in the queue log, Any '|' characters in the caller ID name will
  -     be replaced with '_'. (provided it’s allowed by the existing
  -     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
  -     the Caller ID name is omitted.
  -
  - asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
  -             In cli.conf, you can now define startup commands that run before
  -             core initialization and before module initialization.
  -
  - audiosocket: added support for DTMF frames
  -              The AudioSocket protocol now forwards DTMF frames with payload
  -              type 0x03. The payload is a 1-byte ascii representing the DTMF
  -              digit (0-9,*,#...).
  -
  - Upgrade Notes:
  - --------------
  -
  - ARI: REST over Websocket
  -      This commit adds the ability to make ARI REST requests over the same
  -      websocket used to receive events. See
  -      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
  -
  - Commit Authors:
  - ---------------
  -
  - Albrecht Oster: (1)
  - Alexei Gradinari: (1)
  - Allan Nathanson: (1)
  - Andreas Wehrmann: (1)
  - Ben Ford: (1)
  - Florent CHAUVEAU: (1)
  - George Joseph: (4)
  - Joshua C. Colp: (1)
  - Luz Paz: (1)
  - Mark Murawski: (1)
  - Mike Bradeen: (1)
  - Mkmer: (1)
  - Naveen Albert: (3)
  - Norm Harrison: (2)
  - Peter Jannesen: (1)
  - Phoneben: (1)
  - Sean Bright: (1)
  - Zhai Liangliang: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 19.7.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 19.7.0.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 19.7.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  -
  - Thank you!
  -
  - The following issues are resolved in this release:
  -
  - New Features made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30037] - Add test support to calling external processes (Reported by
  -                    Philip Prindeville)
  - [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
  - [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
  - [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
  -                    key information (Reported by Joshua C. Colp)
  - [ASTERISK-29899] - features: Add advanced transfer initiation options
  -                    (Reported by N A)
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
  -                    variable error (Reported by George Joseph)
  - [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
  -                    geoloc_config.c (Reported by George Joseph)
  - [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
  -                    Fault (Reported by Dan Cropp)
  - [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
  -                    (Reported by sungtae kim)
  - [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
  -                    Dömsödi)
  - [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
  -                    (Reported by N A)
  - [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
  -                    values on incoming channel (Reported by George Joseph)
  - [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
  -                    Alexander Traud)
  - [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
  - [ASTERISK-29453] - alembic: incoming_call_offer_pref and
  -                    outgoing_call_offer_pref missing in "ps_endpoints" table
  -                    (Reported by Daniel Thümen)
  - [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
  -                    (Reported by George Joseph)
  - [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
  - [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
  - [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
  -                    Michael Cargile)
  - [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
  -                    by Michael Cargile)
  - [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
  -                    required parameter. (Reported by Chris Young)
  -
  - Improvements made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
  -                    debugs to DEBUG level (Reported by N A)
  - [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
  -                    Michael Bradeen)
  - [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
  -                    interface to Openssl API's (Reported by Philip Prindeville)
  - [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
  -                    (Reported by Philip Prindeville)
  - [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
  -                    in profiles (Reported by George Joseph)
  - [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
  -                    (Reported by George Joseph)
  - [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
  -                    dynamic configurations (Reported by George Joseph)
  - [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
  -                    (Reported by Alexei Gradinari)
  - [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
  - [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
  -                    A)
  - [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
  - [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
  -                    N A)
  - [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.0.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.0.0.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 20.0.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  -
  - Thank you!
  -
  - The following issues are resolved in this release:
  -
  - Deprecations made in this release:
  - ----------------------------------
  -
  - [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
  -                    N A)
  - [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
  -                    Joshua C. Colp)
  - [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
  -                    C. Colp)
  - [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
  -                    Joshua C. Colp)
  - [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
  -                    C. Colp)
  - [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
  -                    by Joshua C. Colp)
  - [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
  -                    (Reported by Joshua C. Colp)
  - [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
  -                    by Joshua C. Colp)
  - [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
  -                    by Joshua C. Colp)
  - [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
  -                    (Reported by Joshua C. Colp)
  - [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
  -                    (Reported by Joshua C. Colp)
  - [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
  -                    by Joshua C. Colp)
  - [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
  -                    by Joshua C. Colp)
  - [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
  -                    (Reported by Joshua C. Colp)
  -
  - Security bugs fixed in this release:
  - ------------------------------------
  -
  - [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
  -                    Clint Ruoho)
  - [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
  -                    (Reported by Benjamin Keith Ford)
  - [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
  -                    (Reported by Leandro Dardini)
  - [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
  - [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
  -                    user (Reported by Ivan Poddubny)
  -
  - New Features made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30037] - Add test support to calling external processes (Reported by
  -                    Philip Prindeville)
  - [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
  - [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
  - [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
  -                    key information (Reported by Joshua C. Colp)
  - [ASTERISK-29899] - features: Add advanced transfer initiation options
  -                    (Reported by N A)
  - [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
  -                    prefix (Reported by N A)
  - [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
  - [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
  -                    N A)
  - [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
  -                    (Reported by José Lopes)
  - [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
  - [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
  -                    (Reported by N A)
  - [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
  -                    N A)
  - [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
  -                    delay configurable (Reported by N A)
  - [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
  -                    by N A)
  - [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
  -                    but everyone else can (Reported by Michael Cargile)
  - [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
  -                    prefix (Reported by N A)
  - [ASTERISK-29486] - Hint-like extension value lookup function without device
  -                    state (Reported by N A)
  - [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
  -                    A)
  - [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
  - [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
  - [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
  -                    (Reported by N A)
  - [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
  - [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
  - [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
  - [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
  - [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
  - [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
  - [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
  -                    by N A)
  - [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
  -                    Queue Members (Reported by Jamuel Starkey)
  - [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
  - [ASTERISK-29496] - Add SendMF application (Reported by N A)
  - [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
  - [ASTERISK-29628] - Add file and directory functions (Reported by N A)
  - [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
  - [ASTERISK-29546] - Add tone detection module (Reported by N A)
  - [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
  -                    Retji)
  - [ASTERISK-29542] - Add audio scrambler (Reported by N A)
  - [ASTERISK-29478] - Function to drop frames in the TX or RX directions
  -                    (Reported by N A)
  - [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
  -                    (Reported by Igor Goncharovsky)
  - [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
  -                    N A)
  - [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
  -                    variable error (Reported by George Joseph)
  - [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
  -                    geoloc_config.c (Reported by George Joseph)
  - [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
  -                    Fault (Reported by Dan Cropp)
  - [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
  -                    (Reported by sungtae kim)
  - [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
  -                    Dömsödi)
  - [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
  -                    (Reported by N A)
  - [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
  -                    values on incoming channel (Reported by George Joseph)
  - [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
  -                    Alexander Traud)
  - [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
  - [ASTERISK-29453] - alembic: incoming_call_offer_pref and
  -                    outgoing_call_offer_pref missing in "ps_endpoints" table
  -                    (Reported by Daniel Thümen)
  - [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
  -                    Colp)
  - [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
  -                    (Reported by George Joseph)
  - [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
  - [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
  - [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
  -                    Michael Cargile)
  - [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
  -                    by Michael Cargile)
  - [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
  -                    required parameter. (Reported by Chris Young)
  - [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
  -                    A)
  - [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
  -                    (Reported by Sergey V. Lobanov)
  - [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
  -                    (Reported by N A)
  - [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 (Reported by Michael Neuhauser)
  - [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
  -                    (Reported by N A)
  - [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
  -                    (Reported by Sam Banks)
  - [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
  -                    accessing git as root during "make install" (Reported by
  -                    Joshua C. Colp)
  - [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
  -                    with normal endpoints causes infinite loop/crash (Reported by
  -                    N A)
  - [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
  -                    optimization is enabled (Reported by George Joseph)
  - [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
  -                    unsupported at this time (Reported by Morvai Szabolcs)
  - [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
  -                    mandatory (Reported by Dmitry Melekhov)
  - [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
  -                    causes failure (Reported by Kevin Harwell)
  - [ASTERISK-30123] - features: Update automixmon documentation to reflect
  -                    reality (Reported by Trevor Peirce)
  - [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
  -                    Korzun)
  - [ASTERISK-30101] - res_prometheus: Optional load
  -                    res_pjsip_outbound_registration.so (Reported by Boris P.
  -                    Korzun)
  - [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
  -                    (Reported by N A)
  - [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
  -                    removed" (Reported by N A)
  - [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
  -                    console (Reported by N A)
  - [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
  -                    (Reported by N A)
  - [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
  -                    longer work (Reported by N A)
  - [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
  -                    N A)
  - [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
  -                    servers (Reported by Kevin Harwell)
  - [ASTERISK-30097] - console: Recent documentation changes for connecting to
  -                    remote console are inconsistent (Reported by Matthias
  -                    Hensler)
  - [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
  -                    bridge (Reported by Josh Alberts)
  - [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
  -                    is specified in pjsip.conf (Reported by Ray Crumrine)
  - [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
  -                    German (Reported by Christof Efkemann)
  - [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
  -                    (Reported by N A)
  - [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
  -                    unstable system (Reported by N A)
  - [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
  -                    run (Reported by N A)
  - [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
  -                    (Reported by Maximilian Fridrich)
  - [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
  -                    event: dialog XML body (Reported by Marco Paland)
  - [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
  -                    waltermoeller)
  - [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
  - [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
  -                    requests (Reported by LA)
  - [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
  -                    returns a rewritten contact (Reported by Thomas Guebels)
  - [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
  -                    involved on callee disconnect (Reported by N A)
  - [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
  - [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
  -                    available (Reported by Michael Auracher)
  - [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
  -                    Michael Auracher)
  - [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
  -                    Dahdi Call On Hold (Reported by Josh Alberts)
  - [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
  -                    restart (Reported by N A)
  - [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
  -                    missing secrets (Reported by N A)
  - [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
  -                    are always recompiled (Reported by N A)
  - [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
  -                    channel is NULL (Reported by N A)
  - [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
  -                    flags. (documentation bug) (Reported by Rusty Newton)
  - [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
  -                    console colorization" (Reported by Sebastian Gutierrez)
  - [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
  -                    number of groups (Reported by N A)
  - [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
  -                    Petersen)
  - [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
  -                    (Reported by N A)
  - [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
  -                    early_media already enabled (Reported by Mark Petersen)
  - [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
  -                    by Mark Petersen)
  - [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
  -                    A)
  - [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
  - [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
  -                    enabled (Reported by Claude Diderich)
  - [ASTERISK-30006] - res_pjsip: UDP transport does not work when
  -                    async_operations is greater than 1 (Reported by Ross Beer)
  - [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
  -                    keyword (Reported by Jasper Hafkenscheid)
  - [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
  -                    columns (Reported by Gregory Massel)
  - [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
  -                    attributes (Reported by Josh Hogan)
  - [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
  - [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
  -                    Daniel Bonazzi)
  - [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
  -                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
  - [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
  -                    xmlstarlet ro be installed when it shouldn't (Reported by
  -                    George Joseph)
  - [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
  -                    available (Reported by Stefan Ruijsenaars)
  - [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
  -                    printout (Reported by N A)
  - [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
  -                    (Reported by David Herselman)
  - [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
  -                    channel for lack of RTP activity (Reported by Dmitriy Serov)
  - [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
  -                    activity in one way sessions (Reported by Boris P. Korzun)
  - [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
  - [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
  -                    SUBSCRIBE request (Reported by Alexei Gradinari)
  - [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
  -                    (Reported by Arix)
  - [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
  -                    (Reported by Jim Van Meggelen)
  - [ASTERISK-29928] - logging messages truncated when using MUSL runtime
  -                    (Reported by Philip Prindeville)
  - [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
  - [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
  -                    displayed incorreclty (Reported by Jonathan Harris)
  - [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
  -                    error (Reported by Boris P. Korzun)
  - [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
  -                    N A)
  - [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
  -                    Gradinari)
  - [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
  -                    Joshua C. Colp)
  - [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
  - [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
  -                    call event. (Reported by kwk)
  - [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
  -                    by cagdas kopuz)
  - [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
  -                    Mark Petersen)
  - [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
  -                    Yaroslav)
  - [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
  -                    clean up auth_sess (Reported by George Joseph)
  - [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
  - [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
  - [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
  -                    bridges (Reported by Torrey Searle)
  - [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
  -                    failure (Reported by Michał Górny)
  - [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
  -                    qualify_frequency the same value (Reported by Alexei
  -                    Gradinari)
  - [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
  -                    NetBSD (Reported by Michał Górny)
  - [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
  -                    Michał Górny)
  - [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
  -                    Michał Górny)
  - [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
  -                    (Reported by Michał Górny)
  - [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
  -                    code (Reported by N A)
  - [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
  -                    (Reported by Sean Bright)
  - [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
  -                    (Reported by George Joseph)
  - [ASTERISK-29858] - Regression: Using external pjproject not working after
  -                    "hack" commit (Reported by George Joseph)
  - [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
  -                    CALLERID(num) (Reported by Mark Murawski)
  - [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
  - [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
  -                    George Joseph)
  - [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
  -                    hour am/pm (Reported by Vincent Dubois)
  - [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
  -                    Cropp)
  - [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
  -                    Mark Petersen)
  - [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
  -                    in documentation (Reported by Luke Escude)
  - [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
  -                    (Reported by Kevin Harwell)
  - [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
  -                    Davies)
  - [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
  -                    (Reported by Sean Bright)
  - [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
  -                    (Reported by George Joseph)
  - [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
  -                    correct on unhold (Reported by Ross Beer)
  - [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
  -                    channels. (Reported by Krzysztof Trempala)
  - [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
  -                    (Reported by Josh Soref)
  - [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
  -                    some cases (Reported by George Joseph)
  - [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
  -                    specific output dir is set (Reported by Frederic Van Espen)
  - [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
  -                    (Reported by N A)
  - [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
  -                    (Reported by N A)
  - [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
  -                    a coredump (Reported by Mark Petersen)
  - [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
  -                    by Alexander Traud)
  - [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
  -                    (Reported by Joshua C. Colp)
  - [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
  -                    (Reported by Andrew Latham)
  - [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
  -                    (Reported by Alexander Traud)
  - [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
  -                    supply a channel name.." error (Reported by Anil Gupta)
  - [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
  -                    (Reported by Thomas Guebels)
  - [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
  -                    (Reported by Simone Lazzaris)
  - [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
  -                    (Reported by Alexander Traud)
  - [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
  -                    configured and we have a schema warning (Reported by Mario
  -                    Ban)
  - [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
  -                    Traud)
  - [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
  - [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
  -                    Gradinari)
  - [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
  -                    same bridge (Reported by Joshua C. Colp)
  - [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
  - [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
  - [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
  -                    Traud)
  - [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
  - [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
  -                    (Reported by N A)
  - [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
  - [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
  -                    (Reported by Alexander Traud)
  - [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
  -                    dynamic hints (Reported by Daniel Zanutti)
  - [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
  - [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
  -                    (Reported by N A)
  - [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
  -                    (Reported by Alexander Traud)
  - [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
  -                    Alexander Traud)
  - [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
  -                    dispatched. (Reported by Alexander Traud)
  - [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
  -                    by Alexander Traud)
  - [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
  -                    Traud)
  - [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
  -                    George Joseph)
  - [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
  -                    (Reported by George Joseph)
  - [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
  -                    Moritz Fain)
  - [ASTERISK-26497] - make install downloads x86_32 variants of external modules
  -                    on non Intel architectures (Reported by Corey Farrell)
  - [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
  -                    (Reported by Dennis Haney)
  - [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
  -                    (Reported by Michael Munger)
  - [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
  -                    does not support it (Reported by Matthew Kern)
  - [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
  -                    A)
  - [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
  -                    Prescom)
  - [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
  -                    (Reported by Joshua C. Colp)
  - [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
  -                    as dialplan application (Reported by Brian J. Murrell)
  - [ASTERISK-29578] - app_queue: Custom device state using included hints do not
  -                    update (Reported by N A)
  - [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
  -                    Guido Falsi)
  - [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
  -                    attributes (Reported by George Joseph)
  - [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
  -                    by Carlos Oliva)
  - [ASTERISK-29629] - ARI external media channel creation doesn't set option data
  -                    (Reported by sungtae kim)
  - [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
  - [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
  -                    by George Joseph)
  - [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
  -                    files (Reported by Sean Bright)
  - [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
  -                    configured IPv6 servers (Reported by Isaac McDonald)
  - [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
  -                    Alexander Zharov)
  - [ASTERISK-29622] - ARI: external media create doesn't use body parameter
  -                    (Reported by sungtae kim)
  - [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
  -                    (Reported by Alexander Traud)
  - [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
  -                    Mark Murawski)
  - [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
  -                    keepstats=yes (Reported by Luke Escude)
  - [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
  -                    (Reported by Alexander Traud)
  - [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
  -                    MF signaling (Reported by Sarah Autumn)
  - [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
  -                    TRACE (Reported by George Joseph)
  - [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
  -                    timings (Reported by N A)
  - [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
  -                    Tomas Maldonado)
  - [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
  -                    by Alexander Traud)
  - [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
  -                    not defined (happens with OGG/Speex) (Reported by Ernani
  -                    José Camargo Azevedo)
  - [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
  -                    filtering is used (Reported by N A)
  - [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
  -                    Rijnhard Hessel)
  - [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
  - [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
  -                    (Reported by under)
  - [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
  -                    (Reported by Michael Welk)
  -
  - Improvements made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
  -                    debugs to DEBUG level (Reported by N A)
  - [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
  -                    Michael Bradeen)
  - [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
  -                    interface to Openssl API's (Reported by Philip Prindeville)
  - [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
  -                    (Reported by Philip Prindeville)
  - [ASTERISK-30209] - pbx_variables: Use const char for
  -                    pbx_substitute_variables_helper_full_location (Reported by N
  -                    A)
  - [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
  -                    in profiles (Reported by George Joseph)
  - [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
  -                    (Reported by George Joseph)
  - [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
  -                    dynamic configurations (Reported by George Joseph)
  - [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
  -                    (Reported by Alexei Gradinari)
  - [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
  - [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
  -                    A)
  - [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
  - [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
  -                    N A)
  - [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
  - [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
  -                    George Joseph)
  - [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
  -                    by George Joseph)
  - [ASTERISK-30089] - general: fix typos (Reported by N A)
  - [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
  -                    Stanislav Abramenkov)
  - [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
  - [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
  -                    (Reported by Alexei Gradinari)
  - [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
  -                    (Reported by N A)
  - [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
  -                    (Reported by Shloime Rosenblum)
  - [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
  -                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
  -                    Moritz Fain)
  - [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
  -                    registration lapses (Reported by N A)
  - [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
  -                    cadences (Reported by Scott Griepentrog)
  - [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
  - [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
  - [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
  -                    implementation (Reported by Kevin Harwell)
  - [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
  - [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
  -                    by N A)
  - [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
  - [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
  -                    Joseph)
  - [ASTERISK-29976] - Should Readme include information about install_prereq
  -                    script? (Reported by Marcel Wagner)
  - [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
  -                    (Reported by Hugh McMaster)
  - [ASTERISK-25716] - Documentation: Document explanations and examples for
  -                    possible values of DIALSTATUS (Reported by Rusty Newton)
  - [ASTERISK-29980] - build: External binary modules don't use https (Reported by
  -                    INVADE International Ltd.)
  - [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
  - [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
  -                    Kfir Itzhak)
  - [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
  -                    Joseph)
  - [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
  -                    (Reported by N A)
  - [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
  - [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
  - [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
  - [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
  -                    (Reported by N A)
  - [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
  - [ASTERISK-29866] - cli: add core dump information to core show settings
  -                    (Reported by N A)
  - [ASTERISK-29898] - documentation: Add default attributes to documentation
  -                    (Reported by N A)
  - [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
  -                    (Reported by N A)
  - [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
  -                    by N A)
  - [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
  -                    (Reported by Mark Petersen)
  - [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
  -                    announcements (Reported by Mark Petersen)
  - [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
  -                    streams (Reported by Luke Escude)
  - [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
  - [ASTERISK-29848] - documentation: Document special system and channel
  -                    variables (Reported by N A)
  - [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
  -                    by Sean Bright)
  - [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
  - [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
  - [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
  -                    stream (Reported by N A)
  - [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
  -                    CDR and events triggered after the QueueStatus action
  -                    (Reported by Dafi Ni)
  - [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
  -                    calls (Reported by Mark Petersen)
  - [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
  -                    N A)
  - [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
  -                    Petersen)
  - [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
  - [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
  - [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
  -                    with extensions (Reported by N A)
  - [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
  -                    (Reported by Dustin Marquess)
  - [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
  - [ASTERISK-29715] - app_voicemail: Refactor email generation functions
  -                    (Reported by N A)
  - [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
  -                    (Reported by Boris P. Korzun)
  - [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
  - [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
  -                    dial time (Reported by N A)
  - [ASTERISK-29662] - Add mix option to Playback application for say and filename
  -                    (Reported by Shloime Rosenblum)
  - [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
  -                    Rosenblum)
  - [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
  -                    Joseph Nadiv)
  - [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
  -                    by N A)
  - [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
  -                    Korzun)
  - [ASTERISK-29529] - Add custom logging level (Reported by N A)
  - [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
  - [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
  -                    to nonexistent location (Reported by N A)
  - [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
  -                    instructions only when a custom greeting is present (Reported
  -                    by Charlie Smurthwaite)
  - [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
  - [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
  - [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
  -                    cancelled (Reported by N A)
  - [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
  -                    playbacks (Reported by Andre Barbosa)
  - [ASTERISK-29495] - Return integer instead of float if response is a whole
  -                    number (Reported by N A)
  - [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
  - [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
  -                    by N A)
  - [ASTERISK-29528] - Add support for multiple files for agent announcements
  -                    (Reported by N A)
  - [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
  -                    requests (Reported by Sean Bright)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.0.1-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  - The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
  - reported by the community and would have not been possible without your
  - participation. Thank you!
  -
  - The following issue is resolved in this release:
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
  -                  (Reported By: Michael Bradeen)
  - [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
  -                  shawty)
  - [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
  -                  subscription persistence is removed (Reported By: nappsoft)
  - [ASTERISK-30338] Backport 2.13 security fixes from pjproject
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.1.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.1.0.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 20.1.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  -
  - Thank you!
  -
  - The following issues are resolved in this release:
  -
  - Security bugs fixed in this release:
  - ------------------------------------
  -
  - [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
  -                    Benjamin Keith Ford)
  - [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
  -                    (Reported by shawty)
  - [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
  -                    (Reported by Michael Bradeen)
  -
  - Improvements made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30328] - Typo in from_domain description on res_pjsip
  -                    configuration documentation (Reported by Marcel Wagner)
  - [ASTERISK-30316] - res_pjsip: Documentation should point out default if
  -                    contact_user is not being set for outbound registrations
  -                    (Reported by Marcel Wagner)
  - [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
  - [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
  -                    N A)
  - [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
  -                    Caller ID (Reported by N A)
  - [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
  -                    (Reported by N A)
  - [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
  -                    N A)
  - [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
  - [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
  -                    debugs to DEBUG level (Reported by N A)
  - [ASTERISK-30223] - features: add no-answer option to Bridge application
  -                    (Reported by N A)
  - [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
  -                    Maximilian Fridrich)
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
  -                    shouldn't be (Reported by Joshua C. Colp)
  - [ASTERISK-30349] - app_if: Format truncation error (Reported by George
  -                    Joseph)
  - [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
  -                    by Saken)
  - [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
  -                    to user (Reported by N A)
  - [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
  -                    (Reported by N A)
  - [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
  - [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
  -                    by N A)
  - [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
  - [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
  -                    connection terminated and subscription persistence is removed
  -                    (Reported by nappsoft)
  - [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
  -                    wrong stream direction of first call leg (Reported by
  -                    Maximilian Fridrich)
  - [ASTERISK-29998] - sla: deadlock when calling SLAStation application
  -                    (Reported by N A)
  - [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
  -                    George Joseph)
  - [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
  - [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
  -                    asterisk.conf option (Reported by Joshua C. Colp)
  - [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
  -                    A)
  - [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
  -                    (Reported by George Joseph)
  - [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
  -                    logged (Reported by N A)
  - [ASTERISK-30281] - chan_rtp: Local address being used before being set
  -                    (Reported by George Joseph)
  - [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
  -                    stateful response (Reported by Jesse Ross)
  - [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
  -                    MALLOC_DEBUG is enabled (Reported by N A)
  - [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
  -                    Igor Goncharovsky)
  - [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
  -                    N A)
  - [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
  -                    without SDP (Reported by Henning Westerholt)
  - [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
  -                    always cancel all calls (Reported by Frederic LE FOLL)
  - [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
  -                    A)
  - [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
  -                    unregistered, causing crash (Reported by N A)
  - [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
  -                    character not to be spoken (Reported by David Woolley)
  - [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
  -                    correctly (Reported by Philip Prindeville)
  - [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
  -                    A)
  - [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
  -                    channel (Reported by Joeran Vinzens)
  - [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
  -                    Igor Yeroshev)
  - [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
  - [ASTERISK-30243] - func_logic: IF function complains if both branches are
  -                    empty (Reported by N A)
  - [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
  -                    correctly (Reported by Philip Prindeville)
  - [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
  -                    (Reported by N A)
  - [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
  -                    variable error (Reported by George Joseph)
  - [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
  -                    geoloc_config.c (Reported by George Joseph)
  - [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
  -                    directory in /var/lib/asterisk/keys (Reported by George
  -                    Joseph)
  -
  - New Features made in this release:
  - ----------------------------------
  -
  - [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
  -                    (Reported by Matt Jordan)
  - [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
  - [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
  - [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
  -                    (Reported by N A)
  - [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
  -                    (Reported by N A)
  - [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
  -                    done (Reported by N A)
  - [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
  -                    (Reported by N A)
  - [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
  -                    (Reported by N A)
  - [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
  -                    A)
  - [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
  -                    TONE_DETECT (Reported by N A)
  - [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
  -                    (Reported by Maximilian Fridrich)
  - [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
  -                    (Reported by N A)
  - [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
  -                    (Reported by N A)
  - [ASTERISK-29432] - New function to allow access to any channel (Reported by
  -                    N A)
  - [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.2.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.2.0.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 20.2.0 resolves several issues reported by the
  - community and would have not been possible without your participation.
  -
  - Thank you!
  -
  - The following issues are resolved in this release:
  -
  - New Features made in this release:
  - ----------------------------------
  -
  - [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
  -                    N A)
  - [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
  -                    overlap dialing (Reported by N A)
  - [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
  -                    Goncharovsky)
  - [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
  -                    (Reported by N A)
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
  -                    AvayaXAsterisk)
  - [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
  -                    causes jitterbuffer to stall (Reported by N A)
  - [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
  -                    is played (Reported by Jaco Kroon)
  - [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
  -                    autodetected (Reported by Nick French)
  - [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
  -                    (Reported by cmaj)
  - [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
  -                    Ross Beer)
  - [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
  -                    Bright)
  - [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
  -                    Evgrafov)
  - [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
  -                    Sean Bright)
  - [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
  -                    changes (Reported by George Joseph)
  - [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
  -                    (Reported by N A)
  - [ASTERISK-30351] - manager: Originate variables are not added when setvar used
  -                    in manager.conf (Reported by Sebastian Gutierrez)
  - [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
  -                    shouldn't be (Reported by Joshua C. Colp)
  - [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
  -                    TLS-only (Reported by Boris P. Korzun)
  - [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
  -                    (Reported by Sean Bright)
  - [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
  -                    by N A)
  - [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
  -                    callerid supplement executed too late (Reported by Oleg)
  - [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
  -                    moh_passthrough has call on hold (Reported by Benjamin Keith
  -                    Ford)
  - [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
  -                    Michael Bradeen)
  - [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
  -                    by Yury Kirsanov)
  - [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
  -                    when using mixmonitor (Reported by Julien Alie)
  - [ASTERISK-30347] - xmldocs: Remove references to removed applications
  -                    (Reported by N A)
  -
  - Improvements made in this release:
  - ----------------------------------
  -
  - [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
  -                    terminated strings (Reported by Michael Bradeen)
  - [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
  -                    (Reported by Michael Bradeen)
  - [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
  -                    (Reported by Michael Bradeen)
  - [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
  -                    Stanislav Abramenkov)
  - [ASTERISK-30404] - app_directory: Add reading directory configuration from
  -                    custom file (Reported by Michael Bradeen)
  - [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
  -                    JSON_DECODE (Reported by N A)
  - [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
  -                    A)
  - [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
  -                    A)
  - [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
  -                    streams (Reported by George Joseph)
  - [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
  -                    (Reported by N A)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.2.1-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.2.1.
  - This release is available for immediate download at
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The release of Asterisk 20.2.1 resolves several issues reported by the
  - community and would have not been possible without your participation.
  -
  - Thank you!
  -
  - The following issues are resolved in this release:
  -
  - Bugs fixed in this release:
  - ---------------------------
  -
  - [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
  -                    (Reported by N A)
  - [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
  -                    isrl)
  -
  - For a full list of changes in this release, please see the ChangeLog:
  - https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
  -
  - Thank you for your continued support of Asterisk!


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.3.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.3.0.
  -
  - The release artifacts are available for immediate download at 
  - https://github.com/asterisk/asterisk/releases/tag/20.3.0
  - and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release 20.3.0
  - =============================
  -
  - Summary:
  - --------
  -
  - Set up new ChangeLogs directory
  - .github: Add AsteriskReleaser
  - 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)
  - .github: Fix CherryPickTest to only run when it should
  - .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
  - .github: Remove separate set labels step from new PR
  - .github: Refactor CP progress and add new PR test progress
  - res_pjsip: mediasec: Add Security-Client headers after 401
  - .github: Add cherry-pick test progress labels
  - LICENSE: Update link to trademark policy.
  - chan_dahdi: Add dialmode option for FXS lines.
  - .github: Update issue templates
  - .github: Remove unnecessary parameter in CherryPickTest
  - 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.
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  - ### pbx_dundi: Add PJSIP support.
  -     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
  -     pjsip_outgoing_endpoint option to be set in dundi.conf.
  -
  - ### cli: increase channel column width
  -     This change increases the display width on 'core show channels' amd 'core
  -     show channels verbose' 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.
  -
  - ### app_senddtmf: Add SendFlash AMI action.
  -     The SendFlash AMI action now allows sending a hook flash event on a
  -     channel.
  -
  - ### res_http_media_cache: Introduce options and customize
  -     The res_http_media_cache module now attempts to load configuration from the
  -     res_http_media_cache.conf file. The following options were added:
  -      * timeout_secs
  -      * user_agent
  -      * follow_location
  -      * max_redirects
  -      * protocols
  -      * redirect_protocols
  -      * dns_cache_timeout_secs
  -
  - ### test.c: Fix counting of tests and add 2 new tests
  -     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 two new unit tests that just pass and fail to be used
  -     for testing CI itself.
  -
  - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  -     It is now possible to specify the MixMonitorID when calling the manager
  -     action: MixMonitorMute. This will allow an individual MixMonitor instance
  -     to be muted via ID. The MixMonitorID can be stored as a channel variable
  -     using the 'i' MixMonitor option and is returned upon creation if this
  -     option is used. As part of this change, if no MixMonitorID is specified in
  -     the manager action MixMonitorMute, Asterisk will set the mute flag on all
  -     MixMonitor audiohooks on the channel. Previous behavior would set the flag
  -     on the first MixMonitor audiohook found.
  -
  - ### bridge_builtin_features: add beep via touch variable
  -     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
  -     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
  -     will result in a periodic beep being played to the monitored channel upon
  -     MixMontior/Monitor feature start. If an interval less than 5 seconds is
  -     specified, the interval will default to 5 seconds. If the value is set to
  -     an invalid interval, the default of 15 seconds will be used.
  -
  - ### format_sln: add .slin as supported file extension
  -     format_sln now recognizes '.slin' as a valid file extension in addition to
  -     the existing '.sln' and '.raw'.
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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.
  -
  - Closed Issues:
  - --------------
  -
  - #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
  - #39: [Bug]: Remove .gitreview from repository.
  - #43: [Bug]: Link to trademark policy is no longer correct
  - #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
  - #52: [improvement]: Add local optimization begin cel event
  -
  - ### For more details, see:
  - https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.3.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.3.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The following security advisories were resolved in this release:
  - https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
  -
  -
  - Change Log for Release 20.3.1
  - =============================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - apply_patches: Use globbing instead of file/sort.
  - apply_patches: Sort patch list before applying
  - pjsip: Upgrade bundled version to pjproject 2.13.1
  -
  - User Notes:
  - -----------
  -
  - - ### res_http_media_cache: Introduce options and customize
  -   The res_http_media_cache module now attempts to load configuration from the
  -   res_http_media_cache.conf file.
  -   The following options were added:
  -     * timeout_secs
  -     * user_agent
  -     * follow_location
  -     * max_redirects
  -     * protocols
  -     * redirect_protocols
  -     * dns_cache_timeout_secs
  -
  - - ### format_sln: add .slin as supported file extension
  -   format_sln now recognizes '.slin' as a valid file extension in addition to
  -   the existing '.sln' and '.raw'.
  -
  - - ### bridge_builtin_features: add beep via touch variable
  -   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  -   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
  -   seconds will result in a periodic beep being played to the monitored channel
  -   upon MixMontior/Monitor feature start.
  -   If an interval less than 5 seconds is specified, the interval will default
  -   to 5 seconds. If the value is set to an invalid interval, the default of 15
  -   seconds will be used.
  -
  - - ### app_senddtmf: Add SendFlash AMI action.
  -   The SendFlash AMI action now allows sending a hook flash event on a channel.
  -
  - - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  -   It is now possible to specify the MixMonitorID when calling the manager
  -   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
  -   be muted via ID.
  -   The MixMonitorID can be stored as a channel variable using the 'i'
  -   MixMonitor option and is returned upon creation if this option is used.
  -   As part of this change, if no MixMonitorID is specified in the manager
  -   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
  -   audiohooks on the channel. Previous behavior would set the flag on the first
  -   MixMonitor audiohook found.
  -
  - - ### pbx_dundi: Add PJSIP support.
  -   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
  -   pjsip_outgoing_endpoint option to be set in dundi.conf.
  -
  - - ### test.c: Fix counting of tests and add 2 new tests
  -   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 two new unit tests that just pass and fail to be used for testing
  -   CI itself.
  -
  - - ### cli: increase channel column width
  -   This change increases the display width on 'core show channels' amd 'core
  -   show channels verbose'.
  -   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.
  -
  -
  - Upgrade Notes:
  - --------------
  -
  -
  - Closed Issues:
  - --------------
  -
  - #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
  -              applying


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.4.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.4.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  -
  - Change Log for Release 20.4.0
  - =============================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - app.h: Move declaration of ast_getdata_result before its first use
  - doc: Remove obsolete CHANGES-staging and UPGRADE-staging
  - .github: Updates for AsteriskReleaser
  - app_voicemail: fix imap compilation errors
  - res_musiconhold: avoid moh state access on unlocked chan
  - utils: add lock timestamps for DEBUG_THREADS
  - .github: Back out triggering PROpenedOrUpdated by label
  - .github: Move publish docs to new file CreateDocs.yml
  - rest-api: Updates for new documentation site
  - .github: Remove result check from PROpenUpdateGateTests
  - .github: Fix use of 'contains'
  - .github: Add recheck label test to additional jobs
  - .github: Fix recheck label typos
  - .github: Fix recheck label manipulation
  - .github: Allow PR submit checks to be re-run by label
  - 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
  - .github: Rework for merge approval
  - 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
  - .github: Fix issues with cherry-pick-reminder
  - indications: logging changes
  - .github Ignore error when adding reviewrs to PR
  - .github: Update field descriptions for AsteriskReleaser
  - 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.
  - .github: Change title of AsteriskReleaser job
  - asterisk.c: Fix option warning for remote console.
  - .github: Don't add cherry-pick reminder if it's already present
  - .github: Fix quoting in PROpenedOrUpdated
  - .github: Add cherry-pick reminder to new PRs
  - 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.
  - .github: Tweak improvement issue type language.
  - .github: Tweak new feature language, and move feature requests elsewhere.
  - .github: Fix staleness check to only run on certain labels.
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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.
  -
  -
  - Closed Issues:
  - --------------
  -
  - #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
  -             insufficient/ineffective
  - #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
  - #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
  -                     state
  - #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
  - #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
  - #122: [new-feature]: res_musiconhold: Add looplast option
  - #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
  - #174: [bug]: app_voicemail imap compile errors
  - #200: [bug]: Regression: In app.h an enum is used before its declaration.


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.5.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.5.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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."
  - .github: Use generic releaser
  - 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.
  - .github: Fix cherry-pick reminder issues
  - chan_iax2.c: Avoid crash with IAX2 switch support.
  - res_geolocation: Ensure required 'location_info' is present.
  - Adds manager actions to allow move/remove/forward individual messages in a
  - particular mailbox folder. The forward command can be used to copy a message
  - within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  - required to retrieve message ID's.
  - app_voicemail: add CLI commands for message manipulation
  - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid`
  - into the scope of the rtp_instance lock.
  - .github: Minor tweak to Asterisk Releaser
  - .github: Suppress cherry-pick reminder for some situations
  - sig_analog: Allow immediate fake ring to be suppressed.
  -
  - User Notes:
  - -----------
  -
  - ### 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
  -     particular mailbox folder. The forward command can be used to copy a message
  -     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  -     required to retrieve message ID's.
  -     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.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
  -      some resource
  - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
  -       immediate=yes
  - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #181: [improvement]: app_voicemail - add manager actions to display and
  -       manipulate messages
  - #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
  - #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"
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.5.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The following security advisories were resolved in this release:
  - [Path traversal via AMI GetConfig allows access to outside files]
  - (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
  - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
  - initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
  - [PJSIP logging allows attacker to inject fake Asterisk log entries ]
  - (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
  - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
  - 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
  -
  - Change Log for Release asterisk-20.5.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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.
  -
  - User Notes:
  - -----------
  -
  - Upgrade Notes:
  - --------------
  -
  -
  - Closed Issues:
  - --------------


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.5.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.5.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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."
  - .github: Use generic releaser
  - 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.
  - .github: Fix cherry-pick reminder issues
  - chan_iax2.c: Avoid crash with IAX2 switch support.
  - res_geolocation: Ensure required 'location_info' is present.
  - Adds manager actions to allow move/remove/forward individual messages in a
  - particular mailbox folder. The forward command can be used to copy a message
  - within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  - required to retrieve message ID's.
  - app_voicemail: add CLI commands for message manipulation
  - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid`
  - into the scope of the rtp_instance lock.
  - .github: Minor tweak to Asterisk Releaser
  - .github: Suppress cherry-pick reminder for some situations
  - sig_analog: Allow immediate fake ring to be suppressed.
  -
  - User Notes:
  - -----------
  -
  - ### 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
  -     particular mailbox folder. The forward command can be used to copy a message
  -     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  -     required to retrieve message ID's.
  -     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.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
  -      some resource
  - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
  -       immediate=yes
  - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
  - #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
  - #181: [improvement]: app_voicemail - add manager actions to display and
  -       manipulate messages
  - #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
  - #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"
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.6.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.6.0.
  -
  - The release artifacts are available for immediate download at 
  - https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.6.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - 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_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
  - 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
  - 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
  - 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
  - #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
  - #539: [bug]: Existence of logger.xml causes linking failure


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.3.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.3.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.3.0
  - and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release 20.3.0
  - =============================
  -
  - Summary:
  - --------
  -
  - Set up new ChangeLogs directory
  - .github: Add AsteriskReleaser
  - 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)
  - .github: Fix CherryPickTest to only run when it should
  - .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
  - .github: Remove separate set labels step from new PR
  - .github: Refactor CP progress and add new PR test progress
  - res_pjsip: mediasec: Add Security-Client headers after 401
  - .github: Add cherry-pick test progress labels
  - LICENSE: Update link to trademark policy.
  - chan_dahdi: Add dialmode option for FXS lines.
  - .github: Update issue templates
  - .github: Remove unnecessary parameter in CherryPickTest
  - 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.
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  - ### pbx_dundi: Add PJSIP support.
  -     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
  -     pjsip_outgoing_endpoint option to be set in dundi.conf.
  -
  - ### cli: increase channel column width
  -     This change increases the display width on 'core show channels' amd 'core
  -     show channels verbose' 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.
  -
  - ### app_senddtmf: Add SendFlash AMI action.
  -     The SendFlash AMI action now allows sending a hook flash event on a
  -     channel.
  -
  - ### res_http_media_cache: Introduce options and customize
  -     The res_http_media_cache module now attempts to load configuration from the
  -     res_http_media_cache.conf file. The following options were added:
  -      * timeout_secs
  -      * user_agent
  -      * follow_location
  -      * max_redirects
  -      * protocols
  -      * redirect_protocols
  -      * dns_cache_timeout_secs
  -
  - ### test.c: Fix counting of tests and add 2 new tests
  -     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 two new unit tests that just pass and fail to be used
  -     for testing CI itself.
  -
  - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  -     It is now possible to specify the MixMonitorID when calling the manager
  -     action: MixMonitorMute. This will allow an individual MixMonitor instance
  -     to be muted via ID. The MixMonitorID can be stored as a channel variable
  -     using the 'i' MixMonitor option and is returned upon creation if this
  -     option is used. As part of this change, if no MixMonitorID is specified in
  -     the manager action MixMonitorMute, Asterisk will set the mute flag on all
  -     MixMonitor audiohooks on the channel. Previous behavior would set the flag
  -     on the first MixMonitor audiohook found.
  -
  - ### bridge_builtin_features: add beep via touch variable
  -     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
  -     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
  -     will result in a periodic beep being played to the monitored channel upon
  -     MixMontior/Monitor feature start. If an interval less than 5 seconds is
  -     specified, the interval will default to 5 seconds. If the value is set to
  -     an invalid interval, the default of 15 seconds will be used.
  -
  - ### format_sln: add .slin as supported file extension
  -     format_sln now recognizes '.slin' as a valid file extension in addition to
  -     the existing '.sln' and '.raw'.
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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.
  -
  - Closed Issues:
  - --------------
  -
  - #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
  - #39: [Bug]: Remove .gitreview from repository.
  - #43: [Bug]: Link to trademark policy is no longer correct
  - #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
  - #52: [improvement]: Add local optimization begin cel event
  -
  - ### For more details, see:
  - https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.3.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 20.3.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The following security advisories were resolved in this release:
  - https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
  -
  -
  - Change Log for Release 20.3.1
  - =============================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - apply_patches: Use globbing instead of file/sort.
  - apply_patches: Sort patch list before applying
  - pjsip: Upgrade bundled version to pjproject 2.13.1
  -
  - User Notes:
  - -----------
  -
  - - ### res_http_media_cache: Introduce options and customize
  -   The res_http_media_cache module now attempts to load configuration from the
  -   res_http_media_cache.conf file.
  -   The following options were added:
  -     * timeout_secs
  -     * user_agent
  -     * follow_location
  -     * max_redirects
  -     * protocols
  -     * redirect_protocols
  -     * dns_cache_timeout_secs
  -
  - - ### format_sln: add .slin as supported file extension
  -   format_sln now recognizes '.slin' as a valid file extension in addition to
  -   the existing '.sln' and '.raw'.
  -
  - - ### bridge_builtin_features: add beep via touch variable
  -   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
  -   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
  -   seconds will result in a periodic beep being played to the monitored channel
  -   upon MixMontior/Monitor feature start.
  -   If an interval less than 5 seconds is specified, the interval will default
  -   to 5 seconds. If the value is set to an invalid interval, the default of 15
  -   seconds will be used.
  -
  - - ### app_senddtmf: Add SendFlash AMI action.
  -   The SendFlash AMI action now allows sending a hook flash event on a channel.
  -
  - - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
  -   It is now possible to specify the MixMonitorID when calling the manager
  -   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
  -   be muted via ID.
  -   The MixMonitorID can be stored as a channel variable using the 'i'
  -   MixMonitor option and is returned upon creation if this option is used.
  -   As part of this change, if no MixMonitorID is specified in the manager
  -   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
  -   audiohooks on the channel. Previous behavior would set the flag on the first
  -   MixMonitor audiohook found.
  -
  - - ### pbx_dundi: Add PJSIP support.
  -   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
  -   pjsip_outgoing_endpoint option to be set in dundi.conf.
  -
  - - ### test.c: Fix counting of tests and add 2 new tests
  -   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 two new unit tests that just pass and fail to be used for testing
  -   CI itself.
  -
  - - ### cli: increase channel column width
  -   This change increases the display width on 'core show channels' amd 'core
  -   show channels verbose'.
  -   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.
  -
  -
  - Upgrade Notes:
  - --------------
  -
  -
  - Closed Issues:
  - --------------
  -
  - #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
  -              applying


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.4.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of Asterisk
  - 20.4.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  -
  - Change Log for Release 20.4.0
  - =============================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - app.h: Move declaration of ast_getdata_result before its first use
  - doc: Remove obsolete CHANGES-staging and UPGRADE-staging
  - .github: Updates for AsteriskReleaser
  - app_voicemail: fix imap compilation errors
  - res_musiconhold: avoid moh state access on unlocked chan
  - utils: add lock timestamps for DEBUG_THREADS
  - .github: Back out triggering PROpenedOrUpdated by label
  - .github: Move publish docs to new file CreateDocs.yml
  - rest-api: Updates for new documentation site
  - .github: Remove result check from PROpenUpdateGateTests
  - .github: Fix use of 'contains'
  - .github: Add recheck label test to additional jobs
  - .github: Fix recheck label typos
  - .github: Fix recheck label manipulation
  - .github: Allow PR submit checks to be re-run by label
  - 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
  - .github: Rework for merge approval
  - 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
  - .github: Fix issues with cherry-pick-reminder
  - indications: logging changes
  - .github Ignore error when adding reviewrs to PR
  - .github: Update field descriptions for AsteriskReleaser
  - 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.
  - .github: Change title of AsteriskReleaser job
  - asterisk.c: Fix option warning for remote console.
  - .github: Don't add cherry-pick reminder if it's already present
  - .github: Fix quoting in PROpenedOrUpdated
  - .github: Add cherry-pick reminder to new PRs
  - 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.
  - .github: Tweak improvement issue type language.
  - .github: Tweak new feature language, and move feature requests elsewhere.
  - .github: Fix staleness check to only run on certain labels.
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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.
  -
  -
  - Closed Issues:
  - --------------
  -
  - #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
  -             insufficient/ineffective
  - #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
  - #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
  -                     state
  - #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
  - #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
  - #122: [new-feature]: res_musiconhold: Add looplast option
  - #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
  - #174: [bug]: app_voicemail imap compile errors
  - #200: [bug]: Regression: In app.h an enum is used before its declaration.


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.5.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.5.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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."
  - .github: Use generic releaser
  - 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.
  - .github: Fix cherry-pick reminder issues
  - chan_iax2.c: Avoid crash with IAX2 switch support.
  - res_geolocation: Ensure required 'location_info' is present.
  - Adds manager actions to allow move/remove/forward individual messages in a
  - particular mailbox folder. The forward command can be used to copy a message
  - within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  - required to retrieve message ID's.
  - app_voicemail: add CLI commands for message manipulation
  - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid`
  - into the scope of the rtp_instance lock.
  - .github: Minor tweak to Asterisk Releaser
  - .github: Suppress cherry-pick reminder for some situations
  - sig_analog: Allow immediate fake ring to be suppressed.
  -
  - User Notes:
  - -----------
  -
  - ### 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
  -     particular mailbox folder. The forward command can be used to copy a message
  -     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  -     required to retrieve message ID's.
  -     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.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
  -      some resource
  - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
  -       immediate=yes
  - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
  - #179: [bug]: Queue strategy "Linear" with Asterisk 20 on Realtime
  - #181: [improvement]: app_voicemail - add manager actions to display and
  -       manipulate messages
  - #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
  - #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"
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.0.0-1
changelog:
  - The Asterisk Development Team would like to announce 
  - the release of asterisk-21.0.0.
  -
  - The release artifacts are available for immediate download at 
  - https://github.com/asterisk/asterisk/releases/tag/21.0.0
  - and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community 
  - and would have not been possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.0.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
  - [GitHub
  - Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - Update master branch for Asterisk 21
  - translate.c: Prefer better codecs upon translate ties.
  - chan_skinny: Remove deprecated module.
  - app_osplookup: Remove deprecated module.
  - chan_mgcp: Remove deprecated module.
  - chan_alsa: Remove deprecated module.
  - pbx_builtins: Remove deprecated and defunct functionality.
  - chan_sip: Remove deprecated module.
  - app_cdr: Remove deprecated application and option.
  - app_macro: Remove deprecated module.
  - res_monitor: Remove deprecated module.
  - http.c: Minor simplification to HTTP status output.
  - app_osplookup: Remove obsolete sample config.
  - say.c: Fix French time playback. (#42)
  - core: Cleanup gerrit and JIRA references. (#58)
  - utils.h: Deprecate `ast_gethostbyname()`. (#79)
  - res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
  - app_sla: Migrate SLA applications out of app_meetme.
  - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
  - .github: Update AsteriskReleaser for security releases
  - users.conf: Deprecate users.conf configuration.
  - Update version for Asterisk 21
  - Remove unneeded CHANGES and UPGRADE files
  - res_pjsip_pubsub: Add body_type to test_handler for unit tests
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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.
  - 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.
  - Remove unneeded CHANGES and UPGRADE files
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  - ### http.c: Minor simplification to HTTP status output.
  -     For bound addresses, the HTTP status page now combines the bound address
  -     and bound port in a single line. Additionally, the SSL bind address has
  -     been renamed to TLS.
  -
  - Upgrade Notes:
  - --------------
  -
  - ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
  -     ast_gethostbyname() has been deprecated and will be removed in Asterisk
  -     23. New code should use `ast_sockaddr_resolve()` and
  -     `ast_sockaddr_resolve_first_af()`.
  -
  - ### app_sla: Migrate SLA applications out of app_meetme.
  -     The SLAStation and SLATrunk applications have been moved from app_meetme
  -     to app_sla. If you are using these applications and have autoload=no, you
  -     will need to explicitly load this module in modules.conf.
  -
  - ### users.conf: Deprecate users.conf configuration.
  -     The users.conf config is now deprecated and will be removed in a future
  -     version of Asterisk.
  -
  - ### res_monitor: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. This also removes
  -     the 'w' and 'W' options for app_queue. MixMonitor should be default and
  -     only option for all settings that previously used either Monitor or
  -     MixMonitor.
  -
  - ### app_osplookup: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### app_cdr: Remove deprecated application and option.
  -     The previously deprecated NoCDR application has been removed. 
  -     Additionally, the previously deprecated 'e' option to the ResetCDR
  -     application has been removed.
  -
  - ### app_macro: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. For most modules
  -     that interacted with app_macro, this change is limited to no longer looking
  -     for the current context from the macrocontext when set. The following
  -     modules have additional impacts:
  -     app_dial - no longer supports M^ connected/redirecting macro
  -     app_minivm - samples written using macro will no longer work.
  -     The sample needs to be re-written
  -     app_queue - can no longer call a macro on the called party's channel. Use
  -     gosub which is currently supported
  -     ccss - no callback macro, gosub only
  -     app_voicemail - no macro support
  -     channel - remove macrocontext and priority, no connected line or
  -     redirection macro options
  -     options - stdexten is deprecated to gosub as the default and only options
  -     pbx - removed macrolock
  -     pbx_dundi - no longer look for macro
  -     snmp - removed macro context, exten, and priority
  -
  - ### translate.c: Prefer better codecs upon translate ties.
  -     When setting up translation between two codecs the quality was not taken
  -     into account, resulting in suboptimal translation. The quality is now taken
  -     into account, which can reduce the number of translation steps required,
  -     and improve the resulting quality.
  -
  - ### chan_sip: Remove deprecated module.
  -     This module was deprecated in Asterisk 17 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_alsa: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### pbx_builtins: Remove deprecated and defunct functionality.
  -     The previously deprecated ImportVar and SetAMAFlags applications have now
  -     been removed.
  -
  - ### chan_mgcp: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_skinny: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #39: [Bug]: Remove .gitreview from repository.
  - #41: [Bug]: say.c Time announcement does not say o'clock for the French
  -      language
  - #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
  - #78: [improvement]: Deprecate ast_gethostbyname()
  - #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #183: [deprecation]: Deprecate users.conf
  - #226: [improvement]: Apply contact_user to incoming calls
  - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
  - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
  - #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"
  - #263: [bug]: download_externals doesn't always handle versions correctly
  - #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
  - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
  -       build_resource_tree


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.0.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 21.0.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - The following security advisories were resolved in this release:
  - - [Path traversal via AMI GetConfig allows access to outside files]
  -   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
  - - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
  -   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
  - - [PJSIP logging allows attacker to inject fake Asterisk log entries]
  -   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
  - - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
  -   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
  -
  - Change Log for Release asterisk-21.0.1
  - ======================================
  -
  - Links:
  - ------
  -
  - - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
  - - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
  - - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
  - - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - - 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.
  -
  - User Notes:
  - -----------
  -
  - Upgrade Notes:
  - --------------
  -
  - Closed Issues:
  - --------------


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.0.2-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.0.2.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.0.2
  - =======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - - res_rtp_asterisk: Fix regression issues with DTLS client check
  -
  - User Notes:
  - -----------
  -
  - Upgrade Notes:
  - --------------
  -
  - Closed Issues:
  - --------------
  -
  - #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()


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.1.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.1.0.
  -
  - The release artifacts are available for immediate download at 
  - https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.1.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - 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.
  - 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
  - #539: [bug]: Existence of logger.xml causes linking failure


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.0.0-1
changelog:
  - The Asterisk Development Team would like to announce
  - the release of asterisk-21.0.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.0.0
  - and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community
  - and would have not been possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.0.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
  - [GitHub
  - Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Update master branch for Asterisk 21
  - translate.c: Prefer better codecs upon translate ties.
  - chan_skinny: Remove deprecated module.
  - app_osplookup: Remove deprecated module.
  - chan_mgcp: Remove deprecated module.
  - chan_alsa: Remove deprecated module.
  - pbx_builtins: Remove deprecated and defunct functionality.
  - chan_sip: Remove deprecated module.
  - app_cdr: Remove deprecated application and option.
  - app_macro: Remove deprecated module.
  - res_monitor: Remove deprecated module.
  - http.c: Minor simplification to HTTP status output.
  - app_osplookup: Remove obsolete sample config.
  - say.c: Fix French time playback. (#42)
  - core: Cleanup gerrit and JIRA references. (#58)
  - utils.h: Deprecate `ast_gethostbyname()`. (#79)
  - res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
  - app_sla: Migrate SLA applications out of app_meetme.
  - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
  - .github: Update AsteriskReleaser for security releases
  - users.conf: Deprecate users.conf configuration.
  - Update version for Asterisk 21
  - Remove unneeded CHANGES and UPGRADE files
  - res_pjsip_pubsub: Add body_type to test_handler for unit tests
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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.
  - 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.
  - Remove unneeded CHANGES and UPGRADE files
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  - ### http.c: Minor simplification to HTTP status output.
  -     For bound addresses, the HTTP status page now combines the bound address
  -     and bound port in a single line. Additionally, the SSL bind address has
  -     been renamed to TLS.
  -
  - Upgrade Notes:
  - --------------
  -
  - ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
  -     ast_gethostbyname() has been deprecated and will be removed in Asterisk
  -     23. New code should use `ast_sockaddr_resolve()` and
  -     `ast_sockaddr_resolve_first_af()`.
  -
  - ### app_sla: Migrate SLA applications out of app_meetme.
  -     The SLAStation and SLATrunk applications have been moved from app_meetme
  -     to app_sla. If you are using these applications and have autoload=no, you
  -     will need to explicitly load this module in modules.conf.
  -
  - ### users.conf: Deprecate users.conf configuration.
  -     The users.conf config is now deprecated and will be removed in a future
  -     version of Asterisk.
  -
  - ### res_monitor: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. This also removes
  -     the 'w' and 'W' options for app_queue. MixMonitor should be default and
  -     only option for all settings that previously used either Monitor or
  -     MixMonitor.
  -
  - ### app_osplookup: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### app_cdr: Remove deprecated application and option.
  -     The previously deprecated NoCDR application has been removed.
  -     Additionally, the previously deprecated 'e' option to the ResetCDR
  -     application has been removed.
  -
  - ### app_macro: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. For most modules
  -     that interacted with app_macro, this change is limited to no longer looking
  -     for the current context from the macrocontext when set. The following
  -     modules have additional impacts:
  -     app_dial - no longer supports M^ connected/redirecting macro
  -     app_minivm - samples written using macro will no longer work.
  -     The sample needs to be re-written
  -     app_queue - can no longer call a macro on the called party's channel. Use
  -     gosub which is currently supported
  -     ccss - no callback macro, gosub only
  -     app_voicemail - no macro support
  -     channel - remove macrocontext and priority, no connected line or
  -     redirection macro options
  -     options - stdexten is deprecated to gosub as the default and only options
  -     pbx - removed macrolock
  -     pbx_dundi - no longer look for macro
  -     snmp - removed macro context, exten, and priority
  -
  - ### translate.c: Prefer better codecs upon translate ties.
  -     When setting up translation between two codecs the quality was not taken
  -     into account, resulting in suboptimal translation. The quality is now taken
  -     into account, which can reduce the number of translation steps required,
  -     and improve the resulting quality.
  -
  - ### chan_sip: Remove deprecated module.
  -     This module was deprecated in Asterisk 17 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_alsa: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### pbx_builtins: Remove deprecated and defunct functionality.
  -     The previously deprecated ImportVar and SetAMAFlags applications have now
  -     been removed.
  -
  - ### chan_mgcp: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_skinny: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #39: [Bug]: Remove .gitreview from repository.
  - #41: [Bug]: say.c Time announcement does not say o'clock for the French
  -      language
  - #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
  - #78: [improvement]: Deprecate ast_gethostbyname()
  - #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #183: [deprecation]: Deprecate users.conf
  - #226: [improvement]: Apply contact_user to incoming calls
  - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
  - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
  - #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"
  - #263: [bug]: download_externals doesn't always handle versions correctly
  - #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
  - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
  -       build_resource_tree


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.1.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.1.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.1.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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.
  - 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
  - #539: [bug]: Existence of logger.xml causes linking failure


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.2.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.2.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.2.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.2.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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
  - 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.
  - res_pjsip_outbound_registration.c: Add User-Agent header override
  - 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_odbc: remove macrocontext from voicemail_messages table
  - 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.
  -
  - User Notes:
  - -----------

  - ### 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
  -
  - ### res_pjsip_outbound_registration.c: Add User-Agent header override
  -                                        PJSIP outbound registrations now
  -                                        support a per-registration User-Agent
  -                                        header
  -
  - ### 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.
  -
  - ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
  -                         The fix requires removing the macrocontext column from
  -                         the voicemail_messages table in the voicemail database
  -                         via alembic upgrade.
  -
  - ### 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
  -
  - Upgrade Notes:
  - --------------
  -
  - ### 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_voicemail_odbc: remove macrocontext from voicemail_messages table
  -     The fix requires that the voicemail database be upgraded via alembic.
  -     Upgrading to the latest voicemail database via alembic will remove the
  -     macrocontext column from the voicemail_messages table.
  -
  - Closed Issues:
  - --------------
  -
  - #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
  - #351: [improvement]: Refactor res_stir_shaken to use libjwt
  - #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
  - #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
  - #492: [improvement]: res_calendar_icalendar: Print icalendar error if
  -                      available on parsing failure
  - #515: [improvement]: Implement option to override User-Agent-Header on a
  -                      per-registration basis
  - #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
  - #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
  -                      channel frame queue
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.3.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.3.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.3.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.3.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 43
  - Commit Authors: 15
  - Issues Resolved: 26
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip_logger: Preserve logging state on reloads.
  -                        Issuing "pjsip reload" will no longer disable logging
  -                        if it was previously enabled from the CLI.
  -
  - #### loader.c: Allow dependent modules to be unloaded recursively.
  -                In certain circumstances, modules with dependency relations can
  -                have their dependents automatically recursively unloaded and
  -                loaded again using the "module refresh" CLI command or the
  -                ModuleLoad AMI command.
  -
  - #### tcptls/iostream: Add support for setting SNI on client TLS connections
  -                       Secure websocket client connections now send SNI in the
  -                       TLS client hello.
  -
  - #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
  -                                        set identify_by=transport for the pjsip
  -                                        endpoint. Then use the existing 'match'
  -                                        option and the new 'transport' option
  -                                        of the identify. Fixes: #672
  -
  - #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
  -                                        this new feature let users match
  -                                        endpoints based on the indound SIP
  -                                        requests' URI. To do so, add
  -                                        'request_uri' to the endpoint's
  -                                        'identify_by' option. The
  -                                        'match_request_uri' option of the
  -                                        identify can be an exact match for the
  -                                        entire request uri, or a regular
  -                                        expression (between slashes). It's
  -                                        quite similar to the header identifer.
  -                                        Fixes: #599
  -
  - #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
  -                         the GET_TRANSFERRER_DATA dialplan variable can now be
  -                         used also in pjsip.
  -
  - #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
  -                 When using the Originate AMI Action, we now can pass the
  -                 PreDialGoSub parameter, instructing the asterisk to perform an
  -                 subrouting at channel before call start. With this parameter
  -                 an call initiated by AMI can request the channel to start the
  -                 call automaticaly, adding a SIP header to using GoSUB,
  -                 instructing to autoanswer the channel, and proceeding the
  -                 outbuound extension executing. Exemple of an context to
  -                 perform the previus indication:
  -                 [addautoanswer]
  -                 exten => _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
  -                 exten => _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
  -                 exten => _s,n,Return()
  -
  - #### manager.c: Add CLI command to kick AMI sessions.
  -                 The "manager kick session" CLI command now allows kicking a
  -                 specified AMI session.
  -
  - #### chan_dahdi: Allow specifying waitfordialtone per call.
  -                  "waitfordialtone" may now be specified for DAHDI trunk
  -                  channels on a per-call basis using the CHANNEL function.
  -
  - #### Upgrade bundled pjproject to 2.14.1
  -      Bundled pjproject has been upgraded to 2.14.1. For more information visit
  -      pjproject Github page:
  -      https://github.com/pjsip/pjproject/releases/tag/2.14.1
  -
  - 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.
  -
  - Commit Authors:
  - ---------------
  -
  - Fabrice Fontaine: (1)
  - George Joseph: (8)
  - Henrik Liljedahl: (1)
  - Holger Hans Peter Freyther: (1)
  - Ivan Poddubny: (2)
  - Jonatascalebe: (1)
  - Joshua Elson: (1)
  - Martin Nystroem: (1)
  - Martin Tomec: (1)
  - Maximilian Fridrich: (1)
  - Naveen Albert: (14)
  - Sean Bright: (7)
  - Sperl Viktor: (2)
  - Spiridonov Dmitry: (1)
  - Stanislav Abramenkov: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.3.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 21.3.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.3.1
  - and https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.3.1
  -
  - Change Log for Release asterisk-21.3.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
  -    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
  -    requests
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 20.5.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-20.5.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-20.5.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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."
  - .github: Use generic releaser
  - 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.
  - .github: Fix cherry-pick reminder issues
  - chan_iax2.c: Avoid crash with IAX2 switch support.
  - res_geolocation: Ensure required 'location_info' is present.
  - Adds manager actions to allow move/remove/forward individual messages in a
  - particular mailbox folder. The forward command can be used to copy a message
  - within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  - required to retrieve message ID's.
  - app_voicemail: add CLI commands for message manipulation
  - res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid`
  - into the scope of the rtp_instance lock.
  - .github: Minor tweak to Asterisk Releaser
  - .github: Suppress cherry-pick reminder for some situations
  - sig_analog: Allow immediate fake ring to be suppressed.
  -
  - User Notes:
  - -----------
  -
  - ### 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
  -     particular mailbox folder. The forward command can be used to copy a message
  -     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
  -     required to retrieve message ID's.
  -     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.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
  -      some resource
  - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
  -       immediate=yes
  - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #181: [improvement]: app_voicemail - add manager actions to display and
  -       manipulate messages
  - #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
  - #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"
  - #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


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.0.0-1
changelog:
  - The Asterisk Development Team would like to announce
  - the release of asterisk-21.0.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.0.0
  - and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community
  - and would have not been possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.0.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full
  - ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
  - [GitHub
  - Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Update master branch for Asterisk 21
  - translate.c: Prefer better codecs upon translate ties.
  - chan_skinny: Remove deprecated module.
  - app_osplookup: Remove deprecated module.
  - chan_mgcp: Remove deprecated module.
  - chan_alsa: Remove deprecated module.
  - pbx_builtins: Remove deprecated and defunct functionality.
  - chan_sip: Remove deprecated module.
  - app_cdr: Remove deprecated application and option.
  - app_macro: Remove deprecated module.
  - res_monitor: Remove deprecated module.
  - http.c: Minor simplification to HTTP status output.
  - app_osplookup: Remove obsolete sample config.
  - say.c: Fix French time playback. (#42)
  - core: Cleanup gerrit and JIRA references. (#58)
  - utils.h: Deprecate `ast_gethostbyname()`. (#79)
  - res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
  - app_sla: Migrate SLA applications out of app_meetme.
  - rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
  - .github: Update AsteriskReleaser for security releases
  - users.conf: Deprecate users.conf configuration.
  - Update version for Asterisk 21
  - Remove unneeded CHANGES and UPGRADE files
  - res_pjsip_pubsub: Add body_type to test_handler for unit tests
  - 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
  - .github: Update workflow-application-token-action to v2
  - 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.
  - 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.
  - Remove unneeded CHANGES and UPGRADE files
  -
  - User Notes:
  - -----------
  -
  - ### 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.
  -
  - ### http.c: Minor simplification to HTTP status output.
  -     For bound addresses, the HTTP status page now combines the bound address
  -     and bound port in a single line. Additionally, the SSL bind address has
  -     been renamed to TLS.
  -
  - Upgrade Notes:
  - --------------
  -
  - ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
  -     ast_gethostbyname() has been deprecated and will be removed in Asterisk
  -     23. New code should use `ast_sockaddr_resolve()` and
  -     `ast_sockaddr_resolve_first_af()`.
  -
  - ### app_sla: Migrate SLA applications out of app_meetme.
  -     The SLAStation and SLATrunk applications have been moved from app_meetme
  -     to app_sla. If you are using these applications and have autoload=no, you
  -     will need to explicitly load this module in modules.conf.
  -
  - ### users.conf: Deprecate users.conf configuration.
  -     The users.conf config is now deprecated and will be removed in a future
  -     version of Asterisk.
  -
  - ### res_monitor: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. This also removes
  -     the 'w' and 'W' options for app_queue. MixMonitor should be default and
  -     only option for all settings that previously used either Monitor or
  -     MixMonitor.
  -
  - ### app_osplookup: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### app_cdr: Remove deprecated application and option.
  -     The previously deprecated NoCDR application has been removed.
  -     Additionally, the previously deprecated 'e' option to the ResetCDR
  -     application has been removed.
  -
  - ### app_macro: Remove deprecated module.
  -     This module was deprecated in Asterisk 16 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy. For most modules
  -     that interacted with app_macro, this change is limited to no longer looking
  -     for the current context from the macrocontext when set. The following
  -     modules have additional impacts:
  -     app_dial - no longer supports M^ connected/redirecting macro
  -     app_minivm - samples written using macro will no longer work.
  -     The sample needs to be re-written
  -     app_queue - can no longer call a macro on the called party's channel. Use
  -     gosub which is currently supported
  -     ccss - no callback macro, gosub only
  -     app_voicemail - no macro support
  -     channel - remove macrocontext and priority, no connected line or
  -     redirection macro options
  -     options - stdexten is deprecated to gosub as the default and only options
  -     pbx - removed macrolock
  -     pbx_dundi - no longer look for macro
  -     snmp - removed macro context, exten, and priority
  -
  - ### translate.c: Prefer better codecs upon translate ties.
  -     When setting up translation between two codecs the quality was not taken
  -     into account, resulting in suboptimal translation. The quality is now taken
  -     into account, which can reduce the number of translation steps required,
  -     and improve the resulting quality.
  -
  - ### chan_sip: Remove deprecated module.
  -     This module was deprecated in Asterisk 17 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_alsa: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### pbx_builtins: Remove deprecated and defunct functionality.
  -     The previously deprecated ImportVar and SetAMAFlags applications have now
  -     been removed.
  -
  - ### chan_mgcp: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - ### chan_skinny: Remove deprecated module.
  -     This module was deprecated in Asterisk 19 and is now being removed in
  -     accordance with the Asterisk Module Deprecation policy.
  -
  - Closed Issues:
  - --------------
  -
  - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
  -      aarch64 Debian platforms
  - #39: [Bug]: Remove .gitreview from repository.
  - #41: [Bug]: say.c Time announcement does not say o'clock for the French
  -      language
  - #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
  - #78: [improvement]: Deprecate ast_gethostbyname()
  - #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
  - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
  - #183: [deprecation]: Deprecate users.conf
  - #226: [improvement]: Apply contact_user to incoming calls
  - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
  - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
  - #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"
  - #263: [bug]: download_externals doesn't always handle versions correctly
  - #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
  - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
  -       build_resource_tree


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.1.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.1.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.1.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - 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.
  - 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
  - #539: [bug]: Existence of logger.xml causes linking failure


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.4.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.4.0.
  -
  - The release artifacts are available for immediate download at 
  - https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.4.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.4.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
  -
  - Summary:
  - --------
  -
  - Commits: 20
  - Commit Authors: 9
  - Issues Resolved: 8
  - Security Advisories Resolved: 0
  -
  - 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.
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log             
  -                 Add a Queue option log-restricted-caller-id to control whether
  -                 the Restricted Caller ID will be stored in the queue log. If
  -                 log-restricted-caller-id=no then the Caller ID will be stripped
  -                 if the Caller ID is restricted.
  -
  - #### pbx.c: expand fields width of "core show hints"                                 
  -             The fields width of "core show hints" were increased. The width of
  -             "extension" field to 30 characters and the width of the "device
  -             state id" field to 60 characters.
  -
  - #### rtp_engine: add support for multirate RFC2833 digits                           
  -                  No change in configuration is required in order to enable this
  -                  feature. Endpoints configured to use RFC2833 will
  -                  automatically have this enabled. If the endpoint does not
  -                  support this, it should not include it in the SDP
  -                  offer/response.
  -                  Resolves: #699
  -
  - Upgrade Notes:
  - --------------
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log             
  -                 Add a new column to the queues table:
  -                 queue_log_option_log_restricted
  -                 ENUM('0','1','off','on','false','true','no','yes')
  -                 to control whether the Restricted Caller ID will be stored in
  -                 the queue log.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - Bastian Triller: (1)
  - Chrsmj: (1)
  - George Joseph: (4)
  - Igor Goncharovsky: (1)
  - Mike Bradeen: (2)
  - Sean Bright: (7)
  - Tinet-Mucw: (1)
  - Walter Doekes: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.4.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.4.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.4.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.4.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 20
  - Commit Authors: 9
  - Issues Resolved: 8
  - Security Advisories Resolved: 0
  -
  - 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.
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log
  -                 Add a Queue option log-restricted-caller-id to control whether
  -                 the Restricted Caller ID will be stored in the queue log. If
  -                 log-restricted-caller-id=no then the Caller ID will be stripped
  -                 if the Caller ID is restricted.
  -
  - #### pbx.c: expand fields width of "core show hints"
  -             The fields width of "core show hints" were increased. The width of
  -             "extension" field to 30 characters and the width of the "device
  -             state id" field to 60 characters.
  -
  - #### rtp_engine: add support for multirate RFC2833 digits
  -                  No change in configuration is required in order to enable this
  -                  feature. Endpoints configured to use RFC2833 will
  -                  automatically have this enabled. If the endpoint does not
  -                  support this, it should not include it in the SDP
  -                  offer/response.
  -                  Resolves: #699
  -
  - Upgrade Notes:
  - --------------
  -
  - #### app_queue: Add option to not log Restricted Caller ID to queue_log
  -                 Add a new column to the queues table:
  -                 queue_log_option_log_restricted
  -                 ENUM('0','1','off','on','false','true','no','yes')
  -                 to control whether the Restricted Caller ID will be stored in
  -                 the queue log.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - Bastian Triller: (1)
  - Chrsmj: (1)
  - George Joseph: (4)
  - Igor Goncharovsky: (1)
  - Mike Bradeen: (2)
  - Sean Bright: (7)
  - Tinet-Mucw: (1)
  - Walter Doekes: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.4.1-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.4.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.4.1
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.4.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 2
  - Commit Authors: 1
  - Issues Resolved: 2
  - Security Advisories Resolved: 0
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (2)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
  -             "make samples"
  - 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -> 20.9.0
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (2):
  -   voicemail.conf.sample: Fix ':' comment typo
  -   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  -
  - Commit List:
  - ------------
  -
  - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  - voicemail.conf.sample: Fix ':' comment typo
  -
  - Commit Details:
  - ---------------
  -
  - #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
  -                    Author: George Joseph
  -                    Date:   2024-07-25
  -
  -                    There can be empty slots in payload_mapping_tx corresponding
  -                    to dynamic payload types that haven't been seen before so we
  -                    now check for NULL before attempting to use 'type' in the
  -                    call to ast_format_cmp.
  -
  -                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
  -
  -                    Resolves: #822
  -
  - #### voicemail.conf.sample: Fix ':' comment typo
  -                             Author: George Joseph
  -                             Date:   2024-07-24
  -
  -                             ...and removed an errant trailing space.
  -
  -                             Resolves: #819


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.4.2-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 21.4.2.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.4.2
  -
  - Change Log for Release asterisk-21.4.2
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  - - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
  -                        Write=originate, is sufficient permissions for code
  -                        execution / System() dialplan
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
  -                       execution / System() dialplan
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (1):
  -   manager.c: Add entries to Originate blacklist
  -
  - Commit List:
  - ------------
  -
  - manager.c: Add entries to Originate blacklist
  -
  - Commit Details:
  - ---------------
  -
  - #### manager.c: Add entries to Originate blacklist
  -                 Author: George Joseph
  -                 Date:   2024-07-22
  -
  -                 Added Reload and DBdeltree to the list of dialplan application
  -                 that can't be executed via the Originate manager action without
  -                 also having write SYSTEM permissions.
  -
  -                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
  -                 dialplan functions that can't be executed via the Originate
  -                 manager action without also having write SYSTEM permissions.
  -
  -                 If the Queue application is attempted to be run by the
  -                 Originate manager action and an AGI parameter is specified in
  -                 the app data, it'll be rejected unless the manager user has
  -                 either the AGI or SYSTEM permissions.
  -
  -                 Resolves: #GHSA-c4cg-9275-6w44


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.4.3-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 21.4.3.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.4.3
  -
  - Change Log for Release asterisk-21.4.3
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
  -                        A malformed Contact or Record-Route URI in an incoming
  -                        SIP request can cause Asterisk to crash when
  -                        res_resolver_unbound is used
  -
  - Commit Authors:
  - ---------------
  -
  - George Joseph: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
  -                       SIP request can cause Asterisk to crash when
  -                       res_resolver_unbound is used
  -
  - Commits By Author:
  - ------------------
  -
  - #### George Joseph (1):
  -   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -
  - Commit List:
  - ------------
  - res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -
  - Commit Details:
  - ---------------
  -
  - #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
  -                            Author: George Joseph
  -                            Date:   2024-08-12
  -
  -                            The ub_result pointer passed to
  -                            unbound_resolver_callback by libunbound can be NULL
  -                            if the query was for something malformed like `.1`
  -                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
  -                            result and return instead of crashing with a SEGV.
  -                            This causes pjproject to simply cancel the
  -                            transaction with a "No answer record in the DNS
  -                            response" error. The existing "off nominal" unit
  -                            test was also updated to check this condition.
  -
  -                            Although not necessary for this fix, we also made
  -                            ast_dns_resolver_completed() tolerant of a NULL
  -                            result.
  -
  -                            Resolves: GHSA-v428-g3cw-7hv9


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.5.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.5.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.5.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.5.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 24
  - Commit Authors: 8
  - Issues Resolved: 17
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip_notify: add dialplan application
  -                        A new dialplan application PJSIPNotify is now available
  -                        which can send SIP NOTIFY requests from the dialplan.
  -                        The pjsip send notify CLI command has also been enhanced
  -                        to allow sending NOTIFY messages to a specific channel.
  -                        Syntax: pjsip send notify <option> channel <channel>
  -
  - #### channel: Add multi-tenant identifier.
  -               tenantid has been added to channels. It can be read in dialplan
  -               via CHANNEL(tenantid), and it can be set using
  -               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
  -               recommended to use the new tenantid option for pjsip endpoints
  -               (e.g., tenantid=My tenant ID) so that it will show up in
  -               Newchannel events. You can set it like any other channel variable
  -               using set_var in pjsip.conf as well, but note that this will NOT
  -               show up in Newchannel events. Tenant ID is also available in CDR
  -               and can be accessed with CDR(tenantid). The peer tenant ID can
  -               also be accessed with CDR(peertenantid). CEL includes tenant ID
  -               as well if it has been set.
  -
  - #### res_pjsip_config_wizard.c: Refactor load process
  -                                 The res_pjsip_config_wizard.so module can now
  -                                 be reloaded.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### channel: Add multi-tenant identifier.
  -               A new versioned struct (ast_channel_initializers) has been added
  -               that gets passed to __ast_channel_alloc_ap. The new function
  -               ast_channel_alloc_with_initializers should be used when creating
  -               channels that require the use of this struct. Currently the only
  -               value in the struct is for tenantid, but now more fields can be
  -               added to the struct as necessary rather than the
  -               __ast_channel_alloc_ap function. A new option (tenantid) has been
  -               added to endpoints in pjsip.conf as well. CEL has had its version
  -               bumped to include tenant ID.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - Ben Ford: (1)
  - Cade Parker: (1)
  - George Joseph: (11)
  - Jaco Kroon: (1)
  - Mike Bradeen: (3)
  - Sean Bright: (2)
  - Tinet-Mucw: (3)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.6.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.6.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.6.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.6.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 39
  - Commit Authors: 9
  - Issues Resolved: 22
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
  -                 The new "suppress_moh_on_sendonly" endpoint option can be used
  -                 to prevent playing MOH back to a caller if the remote end sends
  -                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
  -
  - #### app_mixmonitor: Add 'D' option for dual-channel audio.
  -                      The MixMonitor application now has a new 'D' option which
  -                      interleaves the recorded audio in the output frames. This
  -                      allows for stereo recording output with one channel being
  -                      the transmitted audio and the other being the received
  -                      audio. The 't' and 't' options are compatible with this.
  -
  - #### manager.c: Restrict ModuleLoad to the configured modules directory.
  -                 The ModuleLoad AMI action now restricts modules to the
  -                 configured modules directory.
  -
  - #### manager: Enhance event filtering for performance
  -               You can now perform more granular filtering on events in
  -               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
  -               = PJSIP/` This is much more efficient than `eventfilter = Event:
  -               Newchannel.*Channel: PJSIP/` Full syntax guide is in
  -               configs/samples/manager.conf.sample.
  -
  - #### db.c: Remove limit on family/key length
  -            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
  -            removed and will now accept families and keys with a total length of
  -            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
  -            dialplan applications, dialplan functions, manager actions and
  -            `databse` CLI commands. Since the media_cache also uses the
  -            `ast_db_*()` APIs, you can now store resources with URIs longer than
  -            253 bytes.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (3)
  - Chrsmj: (1)
  - George Joseph: (15)
  - Jiangxc: (1)
  - Naveen Albert: (7)
  - Peter Jannesen: (2)
  - Sean Bright: (7)
  - Thomas Guebels: (2)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.6.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.6.1
  -
  - Change Log for Release asterisk-21.6.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
  -    Path traversal via AMI ListCategories allows access to outside files
  -
  - User Notes:
  - -----------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory.
  -                 The ListCategories AMI action now restricts files to the
  -                 configured configuration directory.
  -
  - Commit Authors:
  - ---------------
  -
  - Ben Ford: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
  -                         outside files
  -
  - Commits By Author:
  - ------------------
  -
  - #### Ben Ford (1):
  -  - manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit List:
  - ------------
  -
  - manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit Details:
  - ---------------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory.
  -                 Author: Ben Ford
  -                 Date:   2024-12-17
  -
  -                 When using the ListCategories AMI action, it was possible to
  -                 traverse upwards through the directories to files outside of
  -                 the configured configuration directory. This action is now
  -                 restricted to the configured directory and an error will now be
  -                 returned if the specified file is outside of this limitation.
  -
  -                 Resolves: #GHSA-33x6-fj46-6rfh
  -
  - UserNote: The ListCategories AMI action now restricts files to the configured
  -           configuration directory.


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.7.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.7.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.7.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.7.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 53
  - Commit Authors: 20
  - Issues Resolved: 19
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### sig_analog: Add Last Number Redial feature.
  -                  Users can now redial the last number called if the
  -                  lastnumredial setting is set to yes.
  -                  Resolves: #437
  -
  - #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
  -      The SHA-256 and SHA-512-256 algorithms are now available for
  -      authentication as both a UAS and a UAC.
  -
  - #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
  -      Bundled pjproject has been upgraded to 2.15.1. For more information visit
  -      pjproject Github page:
  -      https://github.com/pjsip/pjproject/releases/tag/2.15.1
  -
  - #### res_pjsip: Add new AOR option "qualify_2xx_only"
  -                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
  -                 that can be set so that only 2XX responses to OPTIONS requests
  -                 used to qualify a contact will mark the contact as available.
  -
  - #### app_queue: allow dynamically adding a queue member in paused state.
  -                 Use the p option of AddQueueMember() for paused member state.
  -                 Optionally, use the r(reason) option to specify a custom reason
  -                 for the pause.
  -
  - #### manager.c: Add Processed Call Count to CoreStatus output
  -                 The current processed call count is now returned as
  -                 CoreProcessedCalls from the CoreStatus AMI Action.
  -
  - #### func_curl.c: Add additional CURL options for SSL requests
  -                   The following new configuration options are now available in
  -                   the res_curl.conf file, and the CURL() function:
  -                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
  -                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
  -                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
  -                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
  -                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
  -                   See the libcurl documentation for more details.
  -
  - #### res_stir_shaken: Allow sending Identity headers for unknown TNs
  -                       You can now set the "unknown_tn_attest_level" option in
  -                       the attestation and/or profile objects in
  -                       stir_shaken.conf to enable sending Identity headers for
  -                       callerid TNs not explicitly configured.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### alembic: Database updates required.
  -               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
  -               authentication digest algorithms' 'res_pjsip: Add new AOR option
  -               "qualify_2xx_only"' ...have modified alembic scripts for the
  -               following database tables: ps_aors, ps_contacts, ps_auths,
  -               ps_globals. If you don't use the scripts to update your database,
  -               reads from those tables will succeeed but inserts into the
  -               ps_contacts table by res_pjsip_registrar will fail.
  -
  - Commit Authors:
  - ---------------
  -
  - Abdelkader Boudih: (3)
  - Alexey Khabulyak: (1)
  - Alexey Vasilyev: (1)
  - Allan Nathanson: (2)
  - Artem Umerov: (1)
  - George Joseph: (17)
  - Jaco Kroon: (1)
  - James Terhune: (1)
  - Joshua C. Colp: (1)
  - Kent: (1)
  - Maksim Nesterov: (1)
  - Maximilian Fridrich: (1)
  - Mike Pultz: (3)
  - Naveen Albert: (6)
  - Sean Bright: (6)
  - Sperl Viktor: (2)
  - Stanislav Abramenkov: (2)
  - Steffen Arntz: (1)
  - Tinet-Mucw: (1)
  - Viktor Litvinov: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.8.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.8.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.8.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.8.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - -------
  -
  - Commits: 28
  - Commit Authors: 12
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### ari/pjsip: Make it possible to control transfers through ARI
  -                 Call transfers on the PJSIP channel can now be controlled by
  -                 ARI. This can be enabled by using the
  -                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (1)
  - Fabriziopicconi: (1)
  - George Joseph: (10)
  - Holger Hans Peter Freyther: (1)
  - Jeremy Lainé: (1)
  - Joshua Elson: (1)
  - Luz Paz: (3)
  - Maximilian Fridrich: (1)
  - Mike Bradeen: (1)
  - Naveen Albert: (1)
  - Sean Bright: (6)


author: Ernani Azevedo <azevedo@voipdomain.io> - 21.9.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-21.9.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 21.9.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-21.9.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 24
  - Commit Authors: 18
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
  -                   A Dial timeout on POST /channels/{channelId}/dial will now
  -                   result in a CANCEL and ChannelDestroyed with cause 19 / User
  -                   alerting, no answer. Previously no explicit cause was set,
  -                   resulting in a cause of 16 / Normal Call Clearing.
  -
  - contrib: Add systemd service and timer files for malloc trim.
  -          Service and timer files for systemd have been added to the
  -          contrib/systemd/ directory. If you are experiencing memory issues,
  -          install these files to have "malloc trim" periodically run on the
  -          system.
  -
  - Add log-caller-id-name option to log Caller ID Name in queue log
  -     This patch adds a global configuration option, log-caller-id-name, to
  -     queues.conf to control whether the Caller ID name is logged as parameter 4
  -     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
  -     is included in the queue log, Any '|' characters in the caller ID name will
  -     be replaced with '_'. (provided it’s allowed by the existing
  -     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
  -     the Caller ID name is omitted.
  -
  - asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
  -             In cli.conf, you can now define startup commands that run before
  -             core initialization and before module initialization.
  -
  - audiosocket: added support for DTMF frames
  -              The AudioSocket protocol now forwards DTMF frames with payload
  -              type 0x03. The payload is a 1-byte ascii representing the DTMF
  -              digit (0-9,*,#...).
  -
  - Upgrade Notes:
  - --------------
  -
  - ARI: REST over Websocket
  -      This commit adds the ability to make ARI REST requests over the same
  -      websocket used to receive events. See
  -      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
  -
  - Commit Authors:
  - ---------------
  -
  - Albrecht Oster: (1)
  - Alexei Gradinari: (1)
  - Allan Nathanson: (1)
  - Andreas Wehrmann: (1)
  - Ben Ford: (1)
  - Florent CHAUVEAU: (1)
  - George Joseph: (4)
  - Joshua C. Colp: (1)
  - Luz Paz: (1)
  - Mark Murawski: (1)
  - Mike Bradeen: (1)
  - Mkmer: (1)
  - Naveen Albert: (3)
  - Norm Harrison: (2)
  - Peter Jannesen: (1)
  - Phoneben: (1)
  - Sean Bright: (1)
  - Zhai Liangliang: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.0.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-22.0.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.0.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-22.0.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 22
  - Commit Authors: 8
  - Issues Resolved: 13
  - Security Advisories Resolved: 1
  -  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
  -                         A malformed Contact or Record-Route URI in an incoming
  -                         SIP request can cause Asterisk to crash when
  -                         res_resolver_unbound is used
  -
  - User Notes:
  - -----------
  -
  - #### feat: ARI "ChannelToneDetected" event
  -            Setting the TONE_DETECT dialplan function on a channel in ARI will
  -            now cause a ChannelToneDetected ARI event to be raised when the
  -            specified tone is detected.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### app_record: Add RECORD_TIME output variable.
  -                  The RECORD_TIME variable now contains the duration of Record()
  -                  recordings in milliseconds.
  -
  - Commit Authors:
  - ---------------
  -
  - Alexei Gradinari: (2)
  - George Joseph: (10)
  - Gibbz00: (1)
  - Joshua C. Colp: (2)
  - Mike Bradeen: (2)
  - Naveen Albert: (1)
  - Sean Bright: (2)
  - Tinet-Mucw: (2)


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.1.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-22.1.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.1.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-22.1.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 39
  - Commit Authors: 9
  - Issues Resolved: 22
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
  -                 The new "suppress_moh_on_sendonly" endpoint option can be used
  -                 to prevent playing MOH back to a caller if the remote end sends
  -                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
  -
  - #### app_mixmonitor: Add 'D' option for dual-channel audio.
  -                      The MixMonitor application now has a new 'D' option which
  -                      interleaves the recorded audio in the output frames. This
  -                      allows for stereo recording output with one channel being
  -                      the transmitted audio and the other being the received
  -                      audio. The 't' and 't' options are compatible with this.
  -
  - #### manager.c: Restrict ModuleLoad to the configured modules directory.
  -                 The ModuleLoad AMI action now restricts modules to the
  -                 configured modules directory.
  -
  - #### manager: Enhance event filtering for performance
  -               You can now perform more granular filtering on events in
  -               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
  -               = PJSIP/` This is much more efficient than `eventfilter = Event:
  -               Newchannel.*Channel: PJSIP/` Full syntax guide is in
  -               configs/samples/manager.conf.sample.
  -
  - #### db.c: Remove limit on family/key length
  -            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
  -            removed and will now accept families and keys with a total length of
  -            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
  -            dialplan applications, dialplan functions, manager actions and
  -            `databse` CLI commands. Since the media_cache also uses the
  -            `ast_db_*()` APIs, you can now store resources with URIs longer than
  -            253 bytes.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (3)
  - Chrsmj: (1)
  - George Joseph: (15)
  - Jiangxc: (1)
  - Naveen Albert: (7)
  - Peter Jannesen: (2)
  - Sean Bright: (7)
  - Thomas Guebels: (2)


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.1.1-1
changelog:
  - The Asterisk Development Team would like to announce security release Asterisk
  - 22.1.1.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.1.1
  -
  - Change Log for Release asterisk-22.1.1
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 1
  - Commit Authors: 1
  - Issues Resolved: 0
  - Security Advisories Resolved: 1
  -  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
  -    Path traversal via AMI ListCategories allows access to outside files
  -
  - User Notes:
  - -----------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory
  -                 The ListCategories AMI action now restricts files to the
  -                 configured configuration directory.
  -
  - Commit Authors:
  - ---------------
  -
  - Ben Ford: (1)
  -
  - Issue and Commit Detail:
  - ========================
  -
  - Closed Issues:
  - --------------
  -
  - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
  -                       outside files
  -
  - Commits By Author:
  - ------------------
  -
  - #### Ben Ford (1):
  -  - manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit List:
  - ------------
  -
  - manager.c: Restrict ListCategories to the configuration directory.
  -
  - Commit Details:
  - ---------------
  -
  - #### manager.c: Restrict ListCategories to the configuration directory.
  -                 Author: Ben Ford
  -                 Date:   2024-12-17
  -
  -                 When using the ListCategories AMI action, it was possible to
  -                 traverse upwards through the directories to files outside of
  -                 the configured configuration directory. This action is now
  -                 restricted to the configured directory and an error will now be
  -                 returned if the specified file is outside of this limitation.
  -
  -                 Resolves: #GHSA-33x6-fj46-6rfh
  -
  - UserNote: The ListCategories AMI action now restricts files to the configured
  -           configuration directory.


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.2.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-22.2.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.2.0
  -
  - This release resolves issues reported by the community and would
  - have not been possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-22.2.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 53
  - Commit Authors: 20
  - Issues Resolved: 19
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### sig_analog: Add Last Number Redial feature.
  -                  Users can now redial the last number called if the
  -                  lastnumredial setting is set to yes.
  -                  Resolves: #437
  -
  - #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
  -      The SHA-256 and SHA-512-256 algorithms are now available for
  -      authentication as both a UAS and a UAC.
  -
  - #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
  -      Bundled pjproject has been upgraded to 2.15.1. For more information visit
  -      pjproject Github page:
  -      https://github.com/pjsip/pjproject/releases/tag/2.15.1
  -
  - #### res_pjsip: Add new AOR option "qualify_2xx_only"
  -                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
  -                 that can be set so that only 2XX responses to OPTIONS requests
  -                 used to qualify a contact will mark the contact as available.
  -
  - #### app_queue: allow dynamically adding a queue member in paused state.
  -                 Use the p option of AddQueueMember() for paused member state.
  -                 Optionally, use the r(reason) option to specify a custom reason
  -                 for the pause.
  -
  - #### manager.c: Add Processed Call Count to CoreStatus output
  -                 The current processed call count is now returned as
  -                 CoreProcessedCalls from the CoreStatus AMI Action.
  -
  - #### func_curl.c: Add additional CURL options for SSL requests
  -                   The following new configuration options are now available in
  -                   the res_curl.conf file, and the CURL() function:
  -                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
  -                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
  -                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
  -                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
  -                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
  -                   See the libcurl documentation for more details.
  -
  - #### res_stir_shaken: Allow sending Identity headers for unknown TNs
  -                       You can now set the "unknown_tn_attest_level" option in
  -                       the attestation and/or profile objects in
  -                       stir_shaken.conf to enable sending Identity headers for
  -                       callerid TNs not explicitly configured.
  -
  - Upgrade Notes:
  - --------------
  -
  - #### alembic: Database updates required.
  -               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
  -               authentication digest algorithms' 'res_pjsip: Add new AOR option
  -               "qualify_2xx_only"' ...have modified alembic scripts for the
  -               following database tables: ps_aors, ps_contacts, ps_auths,
  -               ps_globals. If you don't use the scripts to update your database,
  -               reads from those tables will succeeed but inserts into the
  -               ps_contacts table by res_pjsip_registrar will fail.
  -
  - Commit Authors:
  - ---------------
  -
  - Abdelkader Boudih: (3)
  - Alexey Khabulyak: (1)
  - Alexey Vasilyev: (1)
  - Allan Nathanson: (2)
  - Artem Umerov: (1)
  - George Joseph: (17)
  - Jaco Kroon: (1)
  - James Terhune: (1)
  - Joshua C. Colp: (1)
  - Kent: (1)
  - Maksim Nesterov: (1)
  - Maximilian Fridrich: (1)
  - Mike Pultz: (3)
  - Naveen Albert: (6)
  - Sean Bright: (6)
  - Sperl Viktor: (2)
  - Stanislav Abramenkov: (2)
  - Steffen Arntz: (1)
  - Tinet-Mucw: (1)
  - Viktor Litvinov: (1)


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.3.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-22.3.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.3.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-22.3.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 28
  - Commit Authors: 12
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - #### ari/pjsip: Make it possible to control transfers through ARI
  -                 Call transfers on the PJSIP channel can now be controlled by
  -                 ARI. This can be enabled by using the
  -                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
  -
  - Commit Authors:
  - ---------------
  -
  - Allan Nathanson: (1)
  - Ben Ford: (1)
  - Fabriziopicconi: (1)
  - George Joseph: (10)
  - Holger Hans Peter Freyther: (1)
  - Jeremy Lainé: (1)
  - Joshua Elson: (1)
  - Luz Paz: (3)
  - Maximilian Fridrich: (1)
  - Mike Bradeen: (1)
  - Naveen Albert: (1)
  - Sean Bright: (6)


author: Ernani Azevedo <azevedo@voipdomain.io> - 22.4.0-1
changelog:
  - The Asterisk Development Team would like to announce the release of
  - asterisk-22.4.0.
  -
  - The release artifacts are available for immediate download at
  - https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
  - https://downloads.asterisk.org/pub/telephony/asterisk
  -
  - Repository: https://github.com/asterisk/asterisk
  - Tag: 22.4.0
  -
  - This release resolves issues reported by the community and would have not been
  - possible without your participation.
  -
  - Thank You!
  -
  - Change Log for Release asterisk-22.4.0
  - ======================================
  -
  - Links:
  - ------
  -
  - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
  - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
  - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
  - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
  -
  - Summary:
  - --------
  -
  - Commits: 24
  - Commit Authors: 18
  - Issues Resolved: 12
  - Security Advisories Resolved: 0
  -
  - User Notes:
  - -----------
  -
  - stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
  -                   A Dial timeout on POST /channels/{channelId}/dial will now
  -                   result in a CANCEL and ChannelDestroyed with cause 19 / User
  -                   alerting, no answer. Previously no explicit cause was set,
  -                   resulting in a cause of 16 / Normal Call Clearing.
  -
  - contrib: Add systemd service and timer files for malloc trim.
  -          Service and timer files for systemd have been added to the
  -          contrib/systemd/ directory. If you are experiencing memory issues,
  -          install these files to have "malloc trim" periodically run on the
  -          system.
  -
  - Add log-caller-id-name option to log Caller ID Name in queue log
  -     This patch adds a global configuration option, log-caller-id-name, to
  -     queues.conf to control whether the Caller ID name is logged as parameter 4
  -     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
  -     is included in the queue log, Any '|' characters in the caller ID name will
  -     be replaced with '_'. (provided it’s allowed by the existing
  -     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
  -     the Caller ID name is omitted.
  -
  - asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
  -             In cli.conf, you can now define startup commands that run before
  -             core initialization and before module initialization.
  -
  - audiosocket: added support for DTMF frames
  -              The AudioSocket protocol now forwards DTMF frames with payload
  -              type 0x03. The payload is a 1-byte ascii representing the DTMF
  -              digit (0-9,*,#...).
  -
  - Upgrade Notes:
  - --------------
  -
  - ARI: REST over Websocket
  -      This commit adds the ability to make ARI REST requests over the same
  -      websocket used to receive events. See
  -      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
  -
  - Commit Authors:
  - ---------------
  -
  - Albrecht Oster: (1)
  - Alexei Gradinari: (1)
  - Allan Nathanson: (1)
  - Andreas Wehrmann: (1)
  - Ben Ford: (1)
  - Florent CHAUVEAU: (1)
  - George Joseph: (4)
  - Joshua C. Colp: (1)
  - Luz Paz: (1)
  - Mark Murawski: (1)
  - Mike Bradeen: (1)
  - Mkmer: (1)
  - Naveen Albert: (3)
  - Norm Harrison: (2)
  - Peter Jannesen: (1)
  - Phoneben: (1)
  - Sean Bright: (1)
  - Zhai Liangliang: (1)


author: Petr Pisar <ppisar@redhat.com> - 2.11.1-26
changelog:
  - Rename a dependency on libusb-devel (bug #2066607)
  - Move configure to build section because of linker package notes (bug #2045296)
  - Respect distribution CFLAGS
  - Set Perl libdir with a configure option


author: Ernani Azevedo <azevedo@voipdomain.io> - 1.1.21-1
changelog:
  - Update to release 1.1.21
  - Updated actions/checkout to v3 in GitHub Actions CI workflow
  - Silence implicit fallthrough warning on newer GCC versions
  - Added CXXFLAGS="-Wp,-D_GLIBCXX_ASSERTIONS" to gcc-9 and higher builds in GitHub
  -   Actions CI
  - More refinements to GitHub Actions CI
  - Adjust linking order of libmemcached{,util}
  - Change CI to test Ubuntu 18.04 in containers
  - Add gcc-12 build to GitHub Actions CI
  - skip removing entries from sqlite if store-queue-on-shutdown is used
  - add Rocky to supported Linux distributions
  - add loongarch64 support
  - Issue #376: Fix configure error when gperf is absent
  - Add riscv64 support
  - added apt-get for some reqs in ubuntu


author: Miroslav Rezanina <mrezanin@redhat.com> - 2.4-8.el7
changelog:
  - gp-Use-initial-exec-tls-for-libunwind-s-recursion-flag.patch [bz#1339710]
  - Resolves: bz#1339710
    (initalization of 'recursive' tls variable in libunwind stack capturer occasionally triggers deadlock in ceph)


author: Miroslav Rezanina <mrezanin@redhat.com> - 2.4-8.el7
changelog:
  - Rebase to 2.6.1 [bz#1431240]
  - Removed libunwind usage [bz#1467203]
  - Resolves: bz#1431240 
    (gperftools fails to build on s390x, lacks s390x support)
  - Resolves: bz#1467203
    (Please, remove libunwind from the gperftools-libs (and 389-ds-base) requirements)


author: Jerry James <loganjerry@gmail.com> - 2.7.1-11
changelog:
  - Verify the source tarball
  - Stop converting THANKS to UTF-8; it already is
  - Minor spec file cleanups


author: Robert-André Mauchin <zebob.m@gmail.com> - 1.2.0-2
changelog:
  - Fix s390x tests and RISCV build
  - Introduces a patch that fix tests on BigEndian
  - Introduces a patch that disable RVV runtime dispatch for RISCV builds


author: Kevin Fenzi <kevin@scrye.com> - 1.2.0-2
changelog:
  - Enable SSL during install ( thanks Tom Briden <tom.briden@eckoh.com> )
  - Install SSL library and headers ( thanks Tom Briden <tom.briden@eckoh.com> )
  - Fixes rhbz#2266018


author: Remi Collet <remi@remirepo.net> - 2007f-30
changelog:
  - Rebuild for remirepo as dropped from F34
  - drop uw-imap and uw-imap-static and uw-imap-utils
    as only uw-imap-devel and libc-client needed


author: Filip Janus <fjanus@redhat.com> - 5.3.25-57
changelog:
  - Add convert-util subpackage
  - It allowes to convert BerkeleyDB database format to GDBM/LMDB format
  - Disable shiping libdb for rhel except convert-tool


author: David King <amigadave@amigadave.com> - 0.15.1b-27
changelog:
  - Add ID3v1 zero padding patch from Debian
  - Add a fix for CVE-2017-11550 (#1478934)
  - Add a fix for CVE-2004-2779 (#1561983)
  - Use %license, remove Group tag
  - Add BuildRequires on gcc


author: Paul Howarth <paul@city-fan.org> - 1.11.0-1
changelog:
  - Update to 1.11.0 (rhbz#2211200)
    - Adds support for encrypt-then-mac (ETM) MACs
    - Adds support for AES-GCM crypto protocols
    - Adds support for sk-ecdsa-sha2-nistp256 and sk-ssh-ed25519 keys
    - Adds support for RSA certificate authentication
    - Adds FIDO support with *_sk() functions
    - Adds RSA-SHA2 key upgrading to OpenSSL, WinCNG, mbedTLS, OS400 backends
    - Adds Agent Forwarding and libssh2_agent_sign()
    - Adds support for Channel Signal message libssh2_channel_signal_ex()
    - Adds support to get the user auth banner message libssh2_userauth_banner()
    - Adds LIBSSH2_NO_{MD5, HMAC_RIPEMD, DSA, RSA, RSA_SHA1, ECDSA, ED25519,
      AES_CBC, AES_CTR, BLOWFISH, RC4, CAST, 3DES} options
    - Adds direct stream UNIX sockets with libssh2_channel_direct_streamlocal_ex()
    - Adds wolfSSL support to CMake file
    - Adds mbedTLS 3.x support
    - Adds LibreSSL 3.5 support
    - Adds support for CMake "unity" builds
    - Adds CMake support for building shared and static libs in a single pass
    - Adds symbol hiding support to CMake
    - Adds support for libssh2.rc for all build tools
    - Adds .zip, .tar.xz and .tar.bz2 release tarballs
    - Enables ed25519 key support for LibreSSL 3.7.0 or higher
    - Improves OpenSSL 1.1 and 3 compatibility
    - Now requires OpenSSL 1.0.2 or newer
    - Now requires CMake 3.1 or newer
    - SFTP: Adds libssh2_sftp_open_ex_r() and libssh2_sftp_open_r() extended APIs
    - SFTP: No longer has a packet limit when reading a directory
    - SFTP: Now parses attribute extensions if they exist
    - SFTP: No longer will busy loop if SFTP fails to initialize
    - SFTP: Now clear various errors as expected
    - SFTP: No longer skips files if the line buffer is too small
    - SCP: Add option to not quote paths
    - SCP: Enables 64-bit offset support unconditionally
    - Now skips leading \r and \n characters in banner_receive()
    - Enables secure memory zeroing with all build tools on all platforms
    - No longer logs SSH_MSG_REQUEST_FAILURE packets from keepalive
    - Speed up base64 encoding by 7x
    - Assert if there is an attempt to write a value that is too large
    - WinCNG: fix memory leak in _libssh2_dh_secret()
    - Added protection against possible null pointer dereferences
    - Agent now handles overly large comment lengths
    - Now ensure KEX replies don't include extra bytes
    - Fixed possible buffer overflow when receiving SSH_MSG_USERAUTH_BANNER
    - Fixed possible buffer overflow in keyboard interactive code path
    - Fixed overlapping memcpy()
    - Fixed Windows UWP builds
    - Fixed DLL import name
    - Renamed local RANDOM_PADDING macro to avoid unexpected define on Windows
    - Support for building with gcc versions older than 8
    - Improvements to CMake, Makefile, NMakefile, GNUmakefile, autoreconf files
    - Restores ANSI C89 compliance
    - Enabled new compiler warnings and fixed/silenced them
    - Improved error messages
    - Now uses CIFuzz
    - Numerous minor code improvements
    - Improvements to CI builds
    - Improvements to unit tests
    - Improvements to doc files
    - Improvements to example files
    - Removed "old gex" build option
    - Removed no-encryption/no-mac builds
    - Removed support for NetWare and Watcom wmake build files
  - Avoid use of deprecated patch syntax
  - Build static library but don't package it since it's required for the
    test suite (https://github.com/libssh2/libssh2/issues/1056)
  - Remove redundant references to %{_libdir} from pkgconfig file
  - Add patch to work around strict permissions issues for sshd tests


author: Paul Howarth <paul@city-fan.org> - 1.11.1-1
changelog:
  - Update to 1.11.1 (rhbz#2319104)
    - This is an enhancement and bugfix release - see RELEASE_NOTES for details
    - Note also that various algorithms are now deprecated and not built by
      default, which affects this package


author: Davide Cavalca <dcavalca@fedoraproject.org> - 1.47^20180101git9b59468-1
changelog:
  - Update for new versioning policy
  - Various fixes to bring up to date with the packaging guidelines
  - Ensure directories exist before installing


author: Remi Collet <remi@remirepo.net> - 3.2.0-9
changelog:
  - build out of sources tree
  - fix incompatible pointer types using patch from
    https://github.com/php-memcached-dev/php-memcached/pull/555


author: Xavier Bachelot <xavier@bachelot.org> - 19-45
changelog:
  - Add conditional for {pipewire-,}jack-audio-connection-kit-devel
  - Narrow scope of globs for .so/.pc


author: Mamoru TASAKA <mtasaka@fedoraproject.org> - 2.2.0.2-8
changelog:
  - Use %gem_extdir_mri instead of ext for %check due to ruby3.2 change
    for ext cleanup during build


author: Ernani Azevedo <azevedo@voipdomain.io> - 1.6.0-1
changelog:
  - Updated to release 1.6.0
  - capture: added compatibility with openssl >= 3.5.0
  - capture: fixed memory leak while rotating dialogs
  - capture: added save file rotation through SIPHUP signal
  - capture: added support for opening gzip compressed input files
  - hep: received HEP packets can now be saved to PCAP files
  - core: added support for PCRE2 library
  - cf: raw payload preview now properly displays characters near window borders
  - cli: properly display captured dialog count when no interface is used


author: Ernani Azevedo <azevedo@voipdomain.io> - 1.7.0-1
changelog:
  - Updated to release 1.7.0
  - save: add option --text to save captured data to plain text
  - capture: fix memory overflows while parsing IP headers
  - hep: fix hep listener enabled in offline mode
  - core: stop sngrep when parent process has ended
  - ssl: fix decrypt with AES256 GCM SHA384 cipher


author: Ernani Azevedo <azevedo@voipdomain.io> - 1.8.0-1
changelog:
  - Updated to release 1.8.0
  - fix typo in message, thanks to lintian
  - fix compiler warnings about unused variables
  - [fix] debian pkg dependency to libpcre2-8-0
  - Fixed a typo in comment line in filter.c
  - Redefine usage of POSIX signals.
  - Support for building sngrep using CMake


author: Jiri Kucera <jkucera@redhat.com> - 14.4.2.0-20
changelog:
  - added patch that fixes WAV to HCOM conversion abortion on 64 bit big endian
    machines
    + resolves #1558887


author: Jiri Kucera <jkucera@redhat.com> - 14.4.2.0-21
changelog:
  - fix hunks in patches
  - prevents division by zero in src/ao.c
    + fixes/prevents "sox killed by SIGFPE (signal 8)" kind of bugs that appear
      randomly, depending on reporter's HW/environment/OS components
    + related bugs: #1309426, #1226675, #1540762, #1492910


author: Jiri Kucera <jkucera@redhat.com> - 14.4.2.0-22
changelog:
  - added patch that fixes:
    + "divide by zero in startread function in wav.c" (CVE-2017-11332)
    + "invalid memory read in read_samples function in hcom.c" (CVE-2017-11358)
    + "divide by zero in wavwritehdr function in wav.c" (CVE-2017-11359)
    resolves #1480674, #1480675, #1480676, and #1480678


author: Lukáš Tyrychtr <lukastyrychtr@gmail.com> 2.5-9
changelog:
  - Add an obsoletes directive for the old speech-tools-libs as they're
    no longer part of the Festival package.


author: Jonathan Wright <jonathan@almalinux.org> - 1.60.4-1
changelog:
  - update to 1.60.4 rhbz#2334236
  - re-enable builds against libxml2, no more bundled libexpat
  - fixes rhbz#2310136
  - fixes rhbz#2310146
  - fixes rhbz#2310152


author: Dominik Mierzejewski <dominik@greysector.net> - 1.3.0^20230922git9defefa-1
changelog:
  - update to latest git snapshot
  - drop obsolete patch
  - use modern macros
  - run the test suite in %check
  - fixes: CVE-2023-37732 CVE-2023-31975 CVE-2021-33454


